From 0f65a1a9267b8a7ab4678ef20b07532e4c8377ca Mon Sep 17 00:00:00 2001
From: chenlh <2008get@163.com>
Date: 星期二, 10 三月 2026 20:32:24 +0800
Subject: [PATCH] VS版ReverbHallRoom提交
---
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.vcxproj.filters | 93
cbb_RoomReverb/reverb_utils/RandomBuffer.h | 15
cbb_RoomReverb/ReverbHallRoom/pug.txt | 51
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/c9606fc76e9a2740/RANDOMBUFFER.ipch | 0
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/60fac65e2eb3380b/REVERBHALLROOM.ipch | 0
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/DocumentLayout.json | 187
cbb_RoomReverb/reverb_utils/Parameters.h | 326 +
cbb_RoomReverb/reverb_utils/DelayLine.h | 263 +
cbb_RoomReverb/reverb_utils/AllpassDiffuser.h | 149
cbb_RoomReverb/reverb.h | 0
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/FileContentIndex/1593ba8f-bbea-403d-9c37-be218f1cd1ec.vsidx | 0
cbb_RoomReverb/ReverbHallRoom/output.wav | 0
cbb_RoomReverb/reverb_utils/Lp1.h | 86
cbb_RoomReverb/reverb_utils/LcgRandom.h | 76
cbb_RoomReverb/reverb_utils/MultitapDelay.h | 130
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/Browse.VC.db | 0
cbb_RoomReverb/reverb_utils/RandomBuffer.cpp | 36
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.sln | 31
cbb_RoomReverb/reverb_utils/Utils.h | 91
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/FileContentIndex/da58f8b0-b2b4-45b2-8449-0947d4e53dc0.vsidx | 0
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.vcxproj | 161
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/DocumentLayout.backup.json | 187
cbb_RoomReverb/reverb_utils/Biquad.cpp | 226 +
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/FileContentIndex/3a4648d8-dd3f-46ea-993c-503c18760d34.vsidx | 0
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/26c6e6bc21719993/REVERB.ipch | 0
cbb_RoomReverb/reverb_utils/Parameters.cpp | 112
cbb_RoomReverb/reverb_utils/Hp1.h | 91
cbb_RoomReverb/reverb_utils/Biquad.h | 84
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.Build.CppClean.log | 19
cbb_RoomReverb/reverb_utils/ModulatedAllpass.h | 168
cbb_RoomReverb/reverb_utils/ModulatedDelay.h | 107
/dev/null | 2
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/CopilotIndices/17.13.439.2385/SemanticSymbols.db | 0
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/c6c4be2c6e03fbf/REVERB_WRAPPER.ipch | 0
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.cpp | 70
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.vcxproj.FileListAbsolute.txt | 0
cbb_RoomReverb/reverb_utils/Programs.h | 116
cbb_RoomReverb/reverb.cpp | 0
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.vcxproj.user | 4
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.exe.recipe | 11
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/.suo | 0
cbb_RoomReverb/reverb_utils/ReverbController.h | 106
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/FileContentIndex/2575945a-65a1-431d-91d3-3d5f980cae5a.vsidx | 0
cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/CopilotIndices/17.13.439.2385/CodeChunks.db | 0
cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.log | 1
cbb_RoomReverb/reverb_wrapper.c | 73
cbb_RoomReverb/reverb_utils/ReverbChannel.h | 407 ++
cbb_RoomReverb/reverb_wrapper.h | 57
cbb_RoomReverb/dr_wav.h | 8419 ++++++++++++++++++++++++++++++++++++++++++
49 files changed, 11,953 insertions(+), 2 deletions(-)
diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/CopilotIndices/17.13.439.2385/CodeChunks.db b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/CopilotIndices/17.13.439.2385/CodeChunks.db
new file mode 100644
index 0000000..d6893fc
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Binary files differ
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new file mode 100644
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new file mode 100644
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Binary files differ
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new file mode 100644
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diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/FileContentIndex/da58f8b0-b2b4-45b2-8449-0947d4e53dc0.vsidx b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/FileContentIndex/da58f8b0-b2b4-45b2-8449-0947d4e53dc0.vsidx
new file mode 100644
index 0000000..7f322f4
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Binary files differ
diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/.suo b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/.suo
new file mode 100644
index 0000000..3dfff56
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/.suo
Binary files differ
diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/Browse.VC.db b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/Browse.VC.db
new file mode 100644
index 0000000..00bf0a4
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+++ b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/Browse.VC.db
Binary files differ
diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/DocumentLayout.backup.json b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/DocumentLayout.backup.json
new file mode 100644
index 0000000..6d5865f
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diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/26c6e6bc21719993/REVERB.ipch b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/26c6e6bc21719993/REVERB.ipch
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index 0000000..62c56b3
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diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/60fac65e2eb3380b/REVERBHALLROOM.ipch b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/60fac65e2eb3380b/REVERBHALLROOM.ipch
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diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/c6c4be2c6e03fbf/REVERB_WRAPPER.ipch b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/c6c4be2c6e03fbf/REVERB_WRAPPER.ipch
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diff --git a/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/c9606fc76e9a2740/RANDOMBUFFER.ipch b/cbb_RoomReverb/ReverbHallRoom/.vs/ReverbHallRoom/v17/ipch/AutoPCH/c9606fc76e9a2740/RANDOMBUFFER.ipch
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index 0000000..369f4ff
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diff --git a/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.cpp b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.cpp
new file mode 100644
index 0000000..9d8e135
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.cpp
@@ -0,0 +1,70 @@
+锘�+#include <iostream>
+#include <vector>
+#include "reverb.h"
+#include "reverb_wrapper.h"
+#include "dr_wav.h"
+
+
+int main()
+{
+ unsigned int channels = 1, frame_size = 64, sample_rate = 48e3; double dry = 0, early = 0, late = 0.8; int input_mix_on = 0, hight_cut_on = 0; // 1~6
+ int low_cut_on = 0; double input_mix = 0, high_cut_freq = 0, low_cut_freq = 0, cross_seed = 0; int taps_on = 1; double taps_count = 0.5; // 7~12
+ double taps_pre_delay = 0.2, taps_decay = 0.5, taps_length = 0.1; int early_difus_on = 1; double early_difus_count = 0.5, early_difus_delay = 0.3; // 7~19
+ double early_difus_feedback = 0.3, early_difus_mod_amt = 0.2, early_difus_mod_rate = 0.1; int late_mode = 1, late_reflect_on = 1; double late_line_count = 0.3; // 20~25
+ double late_line_size = 0.4, late_line_mod_amt = 0.2, late_line_decay = 0.3, late_line_mod_rate = 0.1, late_difus_count = 0.1, late_difus_delay = 0.2; // 26~31
+ double late_difus_feedback = 0.3, late_difus_mod_amt = 0.3, late_difus_mod_rate = 0.4; int eq_low_shelf_on = 0, eq_high_shelf_on = 0, eq_low_pass_on = 0; // 32~37
+ double eq_low_shelf_freq = 0.1, eq_low_shelf_gain = 0.1, eq_high_shelf_freq = 0.1, eq_high_shelf_gain = 0.1, eq_low_pass_freq = 0.2; // 38~42
+
+ Reverb *reverb[1];
+ reverb_wrapper_init((void **)reverb, channels, frame_size, sample_rate, dry, early, late, input_mix_on, hight_cut_on,
+ low_cut_on, input_mix, high_cut_freq, low_cut_freq, cross_seed, taps_on, taps_count, taps_pre_delay, taps_decay, taps_length, early_difus_on, early_difus_count, early_difus_delay,
+ early_difus_feedback, early_difus_mod_amt, early_difus_mod_rate, late_mode, late_reflect_on, late_line_count, late_line_size, late_line_mod_amt, late_line_decay = 0.1, late_line_mod_rate, late_difus_count, late_difus_delay, // 20~31
+ late_difus_feedback, late_difus_mod_amt, late_difus_mod_rate, eq_low_shelf_on, eq_high_shelf_on, eq_low_pass_on, eq_low_shelf_freq, eq_low_shelf_gain, eq_high_shelf_freq = 0.1, eq_high_shelf_gain, eq_low_pass_freq // 32~42
+ );
+
+
+ // 璇诲叆wav璇煶鏂囦欢
+ drwav_uint64 samples;
+ float* pSampleDataIn = drwav_open_file_and_read_pcm_frames_f32("E:\\music\\Voice_print\\mengwanzhou.wav", &channels, &sample_rate, &samples, NULL);
+ if (pSampleDataIn == NULL) {
+ return -1;
+ }
+
+ int frame_count = samples / frame_size;
+ samples = frame_count * frame_size;
+ std::vector<float> outputData(samples);
+
+ for (int i = 0; i < frame_count; i++) {
+ float *src = pSampleDataIn + i * frame_size;
+ float *dst = outputData.data() + i * frame_size;
+ reverb_wrapper_process(*reverb, src, dst);
+ }
+
+ // 灏嗗鐞嗗悗闊抽鏁版嵁鍐欏叆wav鏂囦欢
+ drwav_data_format format;
+ format.container = drwav_container_riff;
+ format.format = DR_WAVE_FORMAT_IEEE_FLOAT;
+ format.channels = channels;
+ format.sampleRate = sample_rate;
+ format.bitsPerSample = 32;
+
+ drwav wav;
+ if (!drwav_init_file_write(&wav, "output.wav", &format, NULL)) {
+ std::cerr << "Error: Cannot create output file" << std::endl;
+ return -1;
+ }
+
+ drwav_uint64 framesWritten = drwav_write_pcm_frames(&wav, samples, outputData.data());
+ if (framesWritten != samples) {
+ std::cerr << "Error: Did not write all frames." << std::endl;
+ }
+
+ drwav_uninit(&wav);
+ drwav_free(pSampleDataIn, NULL);
+
+ std::cout << "Done." << std::endl;
+
+ return 0;
+}
+
diff --git a/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.sln b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.sln
new file mode 100644
index 0000000..def7780
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.sln
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+EndProject
+Global
+ GlobalSection(SolutionConfigurationPlatforms) = preSolution
+ Debug|x64 = Debug|x64
+ Debug|x86 = Debug|x86
+ Release|x64 = Release|x64
+ Release|x86 = Release|x86
+ EndGlobalSection
+ GlobalSection(ProjectConfigurationPlatforms) = postSolution
+ {1400541B-B713-435F-95FB-9F7F15065495}.Debug|x64.ActiveCfg = Debug|x64
+ {1400541B-B713-435F-95FB-9F7F15065495}.Debug|x64.Build.0 = Debug|x64
+ {1400541B-B713-435F-95FB-9F7F15065495}.Debug|x86.ActiveCfg = Debug|Win32
+ {1400541B-B713-435F-95FB-9F7F15065495}.Debug|x86.Build.0 = Debug|Win32
+ {1400541B-B713-435F-95FB-9F7F15065495}.Release|x64.ActiveCfg = Release|x64
+ {1400541B-B713-435F-95FB-9F7F15065495}.Release|x64.Build.0 = Release|x64
+ {1400541B-B713-435F-95FB-9F7F15065495}.Release|x86.ActiveCfg = Release|Win32
+ {1400541B-B713-435F-95FB-9F7F15065495}.Release|x86.Build.0 = Release|Win32
+ EndGlobalSection
+ GlobalSection(SolutionProperties) = preSolution
+ HideSolutionNode = FALSE
+ EndGlobalSection
+ GlobalSection(ExtensibilityGlobals) = postSolution
+ SolutionGuid = {B1A4C62B-D68D-453D-9503-5D45576018FC}
+ EndGlobalSection
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index 0000000..b054967
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+ </PropertyGroup>
+ <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
+ <ImportGroup Label="ExtensionSettings">
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+ <ClInclude Include="..\reverb_utils\Biquad.h" />
+ <ClInclude Include="..\reverb_utils\DelayLine.h" />
+ <ClInclude Include="..\reverb_utils\Hp1.h" />
+ <ClInclude Include="..\reverb_utils\LcgRandom.h" />
+ <ClInclude Include="..\reverb_utils\Lp1.h" />
+ <ClInclude Include="..\reverb_utils\ModulatedAllpass.h" />
+ <ClInclude Include="..\reverb_utils\ModulatedDelay.h" />
+ <ClInclude Include="..\reverb_utils\MultitapDelay.h" />
+ <ClInclude Include="..\reverb_utils\Parameters.h" />
+ <ClInclude Include="..\reverb_utils\Programs.h" />
+ <ClInclude Include="..\reverb_utils\RandomBuffer.h" />
+ <ClInclude Include="..\reverb_utils\ReverbChannel.h" />
+ <ClInclude Include="..\reverb_utils\ReverbController.h" />
+ <ClInclude Include="..\reverb_utils\Utils.h" />
+ <ClInclude Include="..\reverb_wrapper.h" />
+ </ItemGroup>
+ <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
+ <ImportGroup Label="ExtensionTargets">
+ </ImportGroup>
+</Project>
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new file mode 100644
index 0000000..8c395cb
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+<Project ToolsVersion="4.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
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+ <UniqueIdentifier>{4FC737F1-C7A5-4376-A066-2A32D752A2FF}</UniqueIdentifier>
+ <Extensions>cpp;c;cc;cxx;c++;cppm;ixx;def;odl;idl;hpj;bat;asm;asmx</Extensions>
+ </Filter>
+ <Filter Include="澶存枃浠�>
+ <UniqueIdentifier>{93995380-89BD-4b04-88EB-625FBE52EBFB}</UniqueIdentifier>
+ <Extensions>h;hh;hpp;hxx;h++;hm;inl;inc;ipp;xsd</Extensions>
+ </Filter>
+ <Filter Include="璧勬簮鏂囦欢">
+ <UniqueIdentifier>{67DA6AB6-F800-4c08-8B7A-83BB121AAD01}</UniqueIdentifier>
+ <Extensions>rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav;mfcribbon-ms</Extensions>
+ </Filter>
+ </ItemGroup>
+ <ItemGroup>
+ <ClCompile Include="ReverbHallRoom.cpp">
+ <Filter>婧愭枃浠�/Filter>
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+ <Filter>婧愭枃浠�/Filter>
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+ <ClInclude Include="..\reverb_utils\DelayLine.h">
+ <Filter>婧愭枃浠�/Filter>
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new file mode 100644
index 0000000..88a5509
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom.vcxproj.user
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diff --git a/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.Build.CppClean.log b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.Build.CppClean.log
new file mode 100644
index 0000000..8c037bb
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.Build.CppClean.log
@@ -0,0 +1,19 @@
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\vc143.pdb
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\vc143.idb
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.obj
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\randombuffer.obj
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\parameters.obj
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\biquad.obj
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverb.obj
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverb_wrapper.obj
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\x64\debug\reverbhallroom.exe
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\x64\debug\reverbhallroom.pdb
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.ilk
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\cl.command.1.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\cl.items.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\cl.read.1.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\cl.write.1.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\link.command.1.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\link.read.1.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\link.secondary.1.tlog
+e:\adi_prj\z_git_fold_temp\03-audio_process\cbb_roomreverb\reverbhallroom\reverbhallroom\x64\debug\reverbhallroom.tlog\link.write.1.tlog
diff --git a/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.exe.recipe b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.exe.recipe
new file mode 100644
index 0000000..3bdb75b
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.exe.recipe
@@ -0,0 +1,11 @@
+锘�?xml version="1.0" encoding="utf-8"?>
+<Project>
+ <ProjectOutputs>
+ <ProjectOutput>
+ <FullPath>E:\ADI_Prj\z_git_fold_temp\03-audio_process\cbb_RoomReverb\ReverbHallRoom\x64\Debug\ReverbHallRoom.exe</FullPath>
+ </ProjectOutput>
+ </ProjectOutputs>
+ <ContentFiles />
+ <SatelliteDlls />
+ <NonRecipeFileRefs />
+</Project>
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diff --git a/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.log b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.log
new file mode 100644
index 0000000..5f28270
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+锘�\ No newline at end of file
diff --git a/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.vcxproj.FileListAbsolute.txt b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.vcxproj.FileListAbsolute.txt
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/ReverbHallRoom/x64/Debug/ReverbHallRoom.vcxproj.FileListAbsolute.txt
diff --git a/cbb_RoomReverb/ReverbHallRoom/output.wav b/cbb_RoomReverb/ReverbHallRoom/output.wav
new file mode 100644
index 0000000..4569b7d
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/output.wav
Binary files differ
diff --git a/cbb_RoomReverb/ReverbHallRoom/pug.txt b/cbb_RoomReverb/ReverbHallRoom/pug.txt
new file mode 100644
index 0000000..34a2a5f
--- /dev/null
+++ b/cbb_RoomReverb/ReverbHallRoom/pug.txt
@@ -0,0 +1,51 @@
+FS:48000, CH:1, N:64, Reverb Params:
+Dry: MUTED
+Early: MUTED
+Late: -6.0 dB
+Interpolation: DISABLED
+High_Cut_ON: DISABLED
+Low_Cut_ON: DISABLED
+Input Mix: 0%
+High Cut: 400 Hz
+Low Cut: 20 Hz
+Cross Seed: 0%
+
+Multitap_Delay_ON: ENABLED
+Count: 128
+Pre-delay: 32 ms
+Decay: 50%
+Length: 109 ms
+
+Early_Diffusion_ON: ENABLED
+Diffusion Stages: 6
+Delay: 10 ms
+Feedback: 30%
+Mod Amt: 50%
+Mod Rate: 0.03 Hz
+
+Late_Mode: POST
+Late_Diffusion_ON: ENABLED
+Line_Count: 4
+Line_Size: 72 ms
+Line_Mod_Amt: 50%
+Line_Decay: 109 ms
+Line_Mod_Rate: 0.00 Hz
+Diffusion Stages: 1
+Diffusion_Delay: 28 ms
+Diffusion_Feedback: 30%
+Diffusion_Mod Amt: 75%
+Diffusion_Mod Rate: 0.27 Hz
+
+EQ_Low_Shelf_ON: DISABLED
+EQ_High_Shelf_ON: DISABLED
+EQ_Lowpass_ON: DISABLED
+Low_Freq: 20 Hz
+Low Gain: -18.0 dB
+High Freq: 817 Hz
+High Gain: -18.0 dB
+LP_Cutoff_Freq: 1368 Hz
+
+Tap Seed: 499
+Diffusion Seed: 499
+Delay Seed: 499
+Late Diffusion Seed: 499
diff --git a/cbb_RoomReverb/dr_wav.h b/cbb_RoomReverb/dr_wav.h
new file mode 100644
index 0000000..811ad80
--- /dev/null
+++ b/cbb_RoomReverb/dr_wav.h
@@ -0,0 +1,8419 @@
+
+#ifndef dr_wav_h
+#define dr_wav_h
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define DRWAV_STRINGIFY(x) #x
+#define DRWAV_XSTRINGIFY(x) DRWAV_STRINGIFY(x)
+
+#define DRWAV_VERSION_MAJOR 0
+#define DRWAV_VERSION_MINOR 14
+#define DRWAV_VERSION_REVISION 5
+#define DRWAV_VERSION_STRING DRWAV_XSTRINGIFY(DRWAV_VERSION_MAJOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_MINOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_REVISION)
+
+#include <stddef.h> /* For size_t. */
+
+/* Sized Types */
+typedef signed char drwav_int8;
+typedef unsigned char drwav_uint8;
+typedef signed short drwav_int16;
+typedef unsigned short drwav_uint16;
+typedef signed int drwav_int32;
+typedef unsigned int drwav_uint32;
+#if defined(_MSC_VER) && !defined(__clang__)
+ typedef signed __int64 drwav_int64;
+ typedef unsigned __int64 drwav_uint64;
+#else
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wlong-long"
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc++11-long-long"
+ #endif
+ #endif
+ typedef signed long long drwav_int64;
+ typedef unsigned long long drwav_uint64;
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+ #endif
+#endif
+#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) || defined(__powerpc64__)
+ typedef drwav_uint64 drwav_uintptr;
+#else
+ typedef drwav_uint32 drwav_uintptr;
+#endif
+typedef drwav_uint8 drwav_bool8;
+typedef drwav_uint32 drwav_bool32;
+#define DRWAV_TRUE 1
+#define DRWAV_FALSE 0
+/* End Sized Types */
+
+/* Decorations */
+#if !defined(DRWAV_API)
+ #if defined(DRWAV_DLL)
+ #if defined(_WIN32)
+ #define DRWAV_DLL_IMPORT __declspec(dllimport)
+ #define DRWAV_DLL_EXPORT __declspec(dllexport)
+ #define DRWAV_DLL_PRIVATE static
+ #else
+ #if defined(__GNUC__) && __GNUC__ >= 4
+ #define DRWAV_DLL_IMPORT __attribute__((visibility("default")))
+ #define DRWAV_DLL_EXPORT __attribute__((visibility("default")))
+ #define DRWAV_DLL_PRIVATE __attribute__((visibility("hidden")))
+ #else
+ #define DRWAV_DLL_IMPORT
+ #define DRWAV_DLL_EXPORT
+ #define DRWAV_DLL_PRIVATE static
+ #endif
+ #endif
+
+ #if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION)
+ #define DRWAV_API DRWAV_DLL_EXPORT
+ #else
+ #define DRWAV_API DRWAV_DLL_IMPORT
+ #endif
+ #define DRWAV_PRIVATE DRWAV_DLL_PRIVATE
+ #else
+ #define DRWAV_API extern
+ #define DRWAV_PRIVATE static
+ #endif
+#endif
+/* End Decorations */
+
+/* Result Codes */
+typedef drwav_int32 drwav_result;
+#define DRWAV_SUCCESS 0
+#define DRWAV_ERROR -1 /* A generic error. */
+#define DRWAV_INVALID_ARGS -2
+#define DRWAV_INVALID_OPERATION -3
+#define DRWAV_OUT_OF_MEMORY -4
+#define DRWAV_OUT_OF_RANGE -5
+#define DRWAV_ACCESS_DENIED -6
+#define DRWAV_DOES_NOT_EXIST -7
+#define DRWAV_ALREADY_EXISTS -8
+#define DRWAV_TOO_MANY_OPEN_FILES -9
+#define DRWAV_INVALID_FILE -10
+#define DRWAV_TOO_BIG -11
+#define DRWAV_PATH_TOO_LONG -12
+#define DRWAV_NAME_TOO_LONG -13
+#define DRWAV_NOT_DIRECTORY -14
+#define DRWAV_IS_DIRECTORY -15
+#define DRWAV_DIRECTORY_NOT_EMPTY -16
+#define DRWAV_END_OF_FILE -17
+#define DRWAV_NO_SPACE -18
+#define DRWAV_BUSY -19
+#define DRWAV_IO_ERROR -20
+#define DRWAV_INTERRUPT -21
+#define DRWAV_UNAVAILABLE -22
+#define DRWAV_ALREADY_IN_USE -23
+#define DRWAV_BAD_ADDRESS -24
+#define DRWAV_BAD_SEEK -25
+#define DRWAV_BAD_PIPE -26
+#define DRWAV_DEADLOCK -27
+#define DRWAV_TOO_MANY_LINKS -28
+#define DRWAV_NOT_IMPLEMENTED -29
+#define DRWAV_NO_MESSAGE -30
+#define DRWAV_BAD_MESSAGE -31
+#define DRWAV_NO_DATA_AVAILABLE -32
+#define DRWAV_INVALID_DATA -33
+#define DRWAV_TIMEOUT -34
+#define DRWAV_NO_NETWORK -35
+#define DRWAV_NOT_UNIQUE -36
+#define DRWAV_NOT_SOCKET -37
+#define DRWAV_NO_ADDRESS -38
+#define DRWAV_BAD_PROTOCOL -39
+#define DRWAV_PROTOCOL_UNAVAILABLE -40
+#define DRWAV_PROTOCOL_NOT_SUPPORTED -41
+#define DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED -42
+#define DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED -43
+#define DRWAV_SOCKET_NOT_SUPPORTED -44
+#define DRWAV_CONNECTION_RESET -45
+#define DRWAV_ALREADY_CONNECTED -46
+#define DRWAV_NOT_CONNECTED -47
+#define DRWAV_CONNECTION_REFUSED -48
+#define DRWAV_NO_HOST -49
+#define DRWAV_IN_PROGRESS -50
+#define DRWAV_CANCELLED -51
+#define DRWAV_MEMORY_ALREADY_MAPPED -52
+#define DRWAV_AT_END -53
+/* End Result Codes */
+
+/* Common data formats. */
+#define DR_WAVE_FORMAT_PCM 0x1
+#define DR_WAVE_FORMAT_ADPCM 0x2
+#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3
+#define DR_WAVE_FORMAT_ALAW 0x6
+#define DR_WAVE_FORMAT_MULAW 0x7
+#define DR_WAVE_FORMAT_DVI_ADPCM 0x11
+#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE
+
+/* Flags to pass into drwav_init_ex(), etc. */
+#define DRWAV_SEQUENTIAL 0x00000001
+#define DRWAV_WITH_METADATA 0x00000002
+
+DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision);
+DRWAV_API const char* drwav_version_string(void);
+
+/* Allocation Callbacks */
+typedef struct
+{
+ void* pUserData;
+ void* (* onMalloc)(size_t sz, void* pUserData);
+ void* (* onRealloc)(void* p, size_t sz, void* pUserData);
+ void (* onFree)(void* p, void* pUserData);
+} drwav_allocation_callbacks;
+/* End Allocation Callbacks */
+
+typedef enum
+{
+ DRWAV_SEEK_SET,
+ DRWAV_SEEK_CUR,
+ DRWAV_SEEK_END
+} drwav_seek_origin;
+
+typedef enum
+{
+ drwav_container_riff,
+ drwav_container_rifx,
+ drwav_container_w64,
+ drwav_container_rf64,
+ drwav_container_aiff
+} drwav_container;
+
+typedef struct
+{
+ union
+ {
+ drwav_uint8 fourcc[4];
+ drwav_uint8 guid[16];
+ } id;
+
+ /* The size in bytes of the chunk. */
+ drwav_uint64 sizeInBytes;
+
+ /*
+ RIFF = 2 byte alignment.
+ W64 = 8 byte alignment.
+ */
+ unsigned int paddingSize;
+} drwav_chunk_header;
+
+typedef struct
+{
+ /*
+ The format tag exactly as specified in the wave file's "fmt" chunk. This can be used by applications
+ that require support for data formats not natively supported by dr_wav.
+ */
+ drwav_uint16 formatTag;
+
+ /* The number of channels making up the audio data. When this is set to 1 it is mono, 2 is stereo, etc. */
+ drwav_uint16 channels;
+
+ /* The sample rate. Usually set to something like 44100. */
+ drwav_uint32 sampleRate;
+
+ /* Average bytes per second. You probably don't need this, but it's left here for informational purposes. */
+ drwav_uint32 avgBytesPerSec;
+
+ /* Block align. This is equal to the number of channels * bytes per sample. */
+ drwav_uint16 blockAlign;
+
+ /* Bits per sample. */
+ drwav_uint16 bitsPerSample;
+
+ /* The size of the extended data. Only used internally for validation, but left here for informational purposes. */
+ drwav_uint16 extendedSize;
+
+ /*
+ The number of valid bits per sample. When <formatTag> is equal to WAVE_FORMAT_EXTENSIBLE, <bitsPerSample>
+ is always rounded up to the nearest multiple of 8. This variable contains information about exactly how
+ many bits are valid per sample. Mainly used for informational purposes.
+ */
+ drwav_uint16 validBitsPerSample;
+
+ /* The channel mask. Not used at the moment. */
+ drwav_uint32 channelMask;
+
+ /* The sub-format, exactly as specified by the wave file. */
+ drwav_uint8 subFormat[16];
+} drwav_fmt;
+
+DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT);
+
+
+/*
+Callback for when data is read. Return value is the number of bytes actually read.
+
+pUserData [in] The user data that was passed to drwav_init() and family.
+pBufferOut [out] The output buffer.
+bytesToRead [in] The number of bytes to read.
+
+Returns the number of bytes actually read.
+
+A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until
+either the entire bytesToRead is filled or you have reached the end of the stream.
+*/
+typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
+
+/*
+Callback for when data is written. Returns value is the number of bytes actually written.
+
+pUserData [in] The user data that was passed to drwav_init_write() and family.
+pData [out] A pointer to the data to write.
+bytesToWrite [in] The number of bytes to write.
+
+Returns the number of bytes actually written.
+
+If the return value differs from bytesToWrite, it indicates an error.
+*/
+typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite);
+
+/*
+Callback for when data needs to be seeked.
+
+pUserData [in] The user data that was passed to drwav_init() and family.
+offset [in] The number of bytes to move, relative to the origin. Will never be negative.
+origin [in] The origin of the seek - the current position or the start of the stream.
+
+Returns whether or not the seek was successful.
+
+Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which will be either DRWAV_SEEK_SET or
+DRWAV_SEEK_CUR.
+*/
+typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin);
+
+/*
+Callback for when the current position in the stream needs to be retrieved.
+
+pUserData [in] The user data that was passed to drwav_init() and family.
+pCursor [out] A pointer to a variable to receive the current position in the stream.
+
+Returns whether or not the operation was successful.
+*/
+typedef drwav_bool32 (* drwav_tell_proc)(void* pUserData, drwav_int64* pCursor);
+
+/*
+Callback for when drwav_init_ex() finds a chunk.
+
+pChunkUserData [in] The user data that was passed to the pChunkUserData parameter of drwav_init_ex() and family.
+onRead [in] A pointer to the function to call when reading.
+onSeek [in] A pointer to the function to call when seeking.
+pReadSeekUserData [in] The user data that was passed to the pReadSeekUserData parameter of drwav_init_ex() and family.
+pChunkHeader [in] A pointer to an object containing basic header information about the chunk. Use this to identify the chunk.
+container [in] Whether or not the WAV file is a RIFF or Wave64 container. If you're unsure of the difference, assume RIFF.
+pFMT [in] A pointer to the object containing the contents of the "fmt" chunk.
+
+Returns the number of bytes read + seeked.
+
+To read data from the chunk, call onRead(), passing in pReadSeekUserData as the first parameter. Do the same for seeking with onSeek(). The return value must
+be the total number of bytes you have read _plus_ seeked.
+
+Use the `container` argument to discriminate the fields in `pChunkHeader->id`. If the container is `drwav_container_riff` or `drwav_container_rf64` you should
+use `id.fourcc`, otherwise you should use `id.guid`.
+
+The `pFMT` parameter can be used to determine the data format of the wave file. Use `drwav_fmt_get_format()` to get the sample format, which will be one of the
+`DR_WAVE_FORMAT_*` identifiers.
+
+The read pointer will be sitting on the first byte after the chunk's header. You must not attempt to read beyond the boundary of the chunk.
+*/
+typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader, drwav_container container, const drwav_fmt* pFMT);
+
+
+/* Structure for internal use. Only used for loaders opened with drwav_init_memory(). */
+typedef struct
+{
+ const drwav_uint8* data;
+ size_t dataSize;
+ size_t currentReadPos;
+} drwav__memory_stream;
+
+/* Structure for internal use. Only used for writers opened with drwav_init_memory_write(). */
+typedef struct
+{
+ void** ppData;
+ size_t* pDataSize;
+ size_t dataSize;
+ size_t dataCapacity;
+ size_t currentWritePos;
+} drwav__memory_stream_write;
+
+typedef struct
+{
+ drwav_container container; /* RIFF, W64. */
+ drwav_uint32 format; /* DR_WAVE_FORMAT_* */
+ drwav_uint32 channels;
+ drwav_uint32 sampleRate;
+ drwav_uint32 bitsPerSample;
+} drwav_data_format;
+
+typedef enum
+{
+ drwav_metadata_type_none = 0,
+
+ /*
+ Unknown simply means a chunk that drwav does not handle specifically. You can still ask to
+ receive these chunks as metadata objects. It is then up to you to interpret the chunk's data.
+ You can also write unknown metadata to a wav file. Be careful writing unknown chunks if you
+ have also edited the audio data. The unknown chunks could represent offsets/sizes that no
+ longer correctly correspond to the audio data.
+ */
+ drwav_metadata_type_unknown = 1 << 0,
+
+ /* Only 1 of each of these metadata items are allowed in a wav file. */
+ drwav_metadata_type_smpl = 1 << 1,
+ drwav_metadata_type_inst = 1 << 2,
+ drwav_metadata_type_cue = 1 << 3,
+ drwav_metadata_type_acid = 1 << 4,
+ drwav_metadata_type_bext = 1 << 5,
+
+ /*
+ Wav files often have a LIST chunk. This is a chunk that contains a set of subchunks. For this
+ higher-level metadata API, we don't make a distinction between a regular chunk and a LIST
+ subchunk. Instead, they are all just 'metadata' items.
+
+ There can be multiple of these metadata items in a wav file.
+ */
+ drwav_metadata_type_list_label = 1 << 6,
+ drwav_metadata_type_list_note = 1 << 7,
+ drwav_metadata_type_list_labelled_cue_region = 1 << 8,
+
+ drwav_metadata_type_list_info_software = 1 << 9,
+ drwav_metadata_type_list_info_copyright = 1 << 10,
+ drwav_metadata_type_list_info_title = 1 << 11,
+ drwav_metadata_type_list_info_artist = 1 << 12,
+ drwav_metadata_type_list_info_comment = 1 << 13,
+ drwav_metadata_type_list_info_date = 1 << 14,
+ drwav_metadata_type_list_info_genre = 1 << 15,
+ drwav_metadata_type_list_info_album = 1 << 16,
+ drwav_metadata_type_list_info_tracknumber = 1 << 17,
+ drwav_metadata_type_list_info_location = 1 << 18,
+ drwav_metadata_type_list_info_organization = 1 << 19,
+ drwav_metadata_type_list_info_keywords = 1 << 20,
+ drwav_metadata_type_list_info_medium = 1 << 21,
+ drwav_metadata_type_list_info_description = 1 << 22,
+
+ /* Other type constants for convenience. */
+ drwav_metadata_type_list_all_info_strings = drwav_metadata_type_list_info_software
+ | drwav_metadata_type_list_info_copyright
+ | drwav_metadata_type_list_info_title
+ | drwav_metadata_type_list_info_artist
+ | drwav_metadata_type_list_info_comment
+ | drwav_metadata_type_list_info_date
+ | drwav_metadata_type_list_info_genre
+ | drwav_metadata_type_list_info_album
+ | drwav_metadata_type_list_info_tracknumber
+ | drwav_metadata_type_list_info_location
+ | drwav_metadata_type_list_info_organization
+ | drwav_metadata_type_list_info_keywords
+ | drwav_metadata_type_list_info_medium
+ | drwav_metadata_type_list_info_description,
+
+ drwav_metadata_type_list_all_adtl = drwav_metadata_type_list_label
+ | drwav_metadata_type_list_note
+ | drwav_metadata_type_list_labelled_cue_region,
+
+ drwav_metadata_type_all = -2, /*0xFFFFFFFF & ~drwav_metadata_type_unknown,*/
+ drwav_metadata_type_all_including_unknown = -1 /*0xFFFFFFFF,*/
+} drwav_metadata_type;
+
+/*
+Sampler Metadata
+
+The sampler chunk contains information about how a sound should be played in the context of a whole
+audio production, and when used in a sampler. See https://en.wikipedia.org/wiki/Sample-based_synthesis.
+*/
+typedef enum
+{
+ drwav_smpl_loop_type_forward = 0,
+ drwav_smpl_loop_type_pingpong = 1,
+ drwav_smpl_loop_type_backward = 2
+} drwav_smpl_loop_type;
+
+typedef struct
+{
+ /* The ID of the associated cue point, see drwav_cue and drwav_cue_point. As with all cue point IDs, this can correspond to a label chunk to give this loop a name, see drwav_list_label_or_note. */
+ drwav_uint32 cuePointId;
+
+ /* See drwav_smpl_loop_type. */
+ drwav_uint32 type;
+
+ /* The offset of the first sample to be played in the loop. */
+ drwav_uint32 firstSampleOffset;
+
+ /* The offset into the audio data of the last sample to be played in the loop. */
+ drwav_uint32 lastSampleOffset;
+
+ /* A value to represent that playback should occur at a point between samples. This value ranges from 0 to UINT32_MAX. Where a value of 0 means no fraction, and a value of (UINT32_MAX / 2) would mean half a sample. */
+ drwav_uint32 sampleFraction;
+
+ /* Number of times to play the loop. 0 means loop infinitely. */
+ drwav_uint32 playCount;
+} drwav_smpl_loop;
+
+typedef struct
+{
+ /* IDs for a particular MIDI manufacturer. 0 if not used. */
+ drwav_uint32 manufacturerId;
+ drwav_uint32 productId;
+
+ /* The period of 1 sample in nanoseconds. */
+ drwav_uint32 samplePeriodNanoseconds;
+
+ /* The MIDI root note of this file. 0 to 127. */
+ drwav_uint32 midiUnityNote;
+
+ /* The fraction of a semitone up from the given MIDI note. This is a value from 0 to UINT32_MAX, where 0 means no change and (UINT32_MAX / 2) is half a semitone (AKA 50 cents). */
+ drwav_uint32 midiPitchFraction;
+
+ /* Data relating to SMPTE standards which are used for syncing audio and video. 0 if not used. */
+ drwav_uint32 smpteFormat;
+ drwav_uint32 smpteOffset;
+
+ /* drwav_smpl_loop loops. */
+ drwav_uint32 sampleLoopCount;
+
+ /* Optional sampler-specific data. */
+ drwav_uint32 samplerSpecificDataSizeInBytes;
+
+ drwav_smpl_loop* pLoops;
+ drwav_uint8* pSamplerSpecificData;
+} drwav_smpl;
+
+/*
+Instrument Metadata
+
+The inst metadata contains data about how a sound should be played as part of an instrument. This
+commonly read by samplers. See https://en.wikipedia.org/wiki/Sample-based_synthesis.
+*/
+typedef struct
+{
+ drwav_int8 midiUnityNote; /* The root note of the audio as a MIDI note number. 0 to 127. */
+ drwav_int8 fineTuneCents; /* -50 to +50 */
+ drwav_int8 gainDecibels; /* -64 to +64 */
+ drwav_int8 lowNote; /* 0 to 127 */
+ drwav_int8 highNote; /* 0 to 127 */
+ drwav_int8 lowVelocity; /* 1 to 127 */
+ drwav_int8 highVelocity; /* 1 to 127 */
+} drwav_inst;
+
+/*
+Cue Metadata
+
+Cue points are markers at specific points in the audio. They often come with an associated piece of
+drwav_list_label_or_note metadata which contains the text for the marker.
+*/
+typedef struct
+{
+ /* Unique identification value. */
+ drwav_uint32 id;
+
+ /* Set to 0. This is only relevant if there is a 'playlist' chunk - which is not supported by dr_wav. */
+ drwav_uint32 playOrderPosition;
+
+ /* Should always be "data". This represents the fourcc value of the chunk that this cue point corresponds to. dr_wav only supports a single data chunk so this should always be "data". */
+ drwav_uint8 dataChunkId[4];
+
+ /* Set to 0. This is only relevant if there is a wave list chunk. dr_wav, like lots of readers/writers, do not support this. */
+ drwav_uint32 chunkStart;
+
+ /* Set to 0 for uncompressed formats. Else the last byte in compressed wave data where decompression can begin to find the value of the corresponding sample value. */
+ drwav_uint32 blockStart;
+
+ /* For uncompressed formats this is the offset of the cue point into the audio data. For compressed formats this is relative to the block specified with blockStart. */
+ drwav_uint32 sampleOffset;
+} drwav_cue_point;
+
+typedef struct
+{
+ drwav_uint32 cuePointCount;
+ drwav_cue_point *pCuePoints;
+} drwav_cue;
+
+/*
+Acid Metadata
+
+This chunk contains some information about the time signature and the tempo of the audio.
+*/
+typedef enum
+{
+ drwav_acid_flag_one_shot = 1, /* If this is not set, then it is a loop instead of a one-shot. */
+ drwav_acid_flag_root_note_set = 2,
+ drwav_acid_flag_stretch = 4,
+ drwav_acid_flag_disk_based = 8,
+ drwav_acid_flag_acidizer = 16 /* Not sure what this means. */
+} drwav_acid_flag;
+
+typedef struct
+{
+ /* A bit-field, see drwav_acid_flag. */
+ drwav_uint32 flags;
+
+ /* Valid if flags contains drwav_acid_flag_root_note_set. It represents the MIDI root note the file - a value from 0 to 127. */
+ drwav_uint16 midiUnityNote;
+
+ /* Reserved values that should probably be ignored. reserved1 seems to often be 128 and reserved2 is 0. */
+ drwav_uint16 reserved1;
+ float reserved2;
+
+ /* Number of beats. */
+ drwav_uint32 numBeats;
+
+ /* The time signature of the audio. */
+ drwav_uint16 meterDenominator;
+ drwav_uint16 meterNumerator;
+
+ /* Beats per minute of the track. Setting a value of 0 suggests that there is no tempo. */
+ float tempo;
+} drwav_acid;
+
+/*
+Cue Label or Note metadata
+
+These are 2 different types of metadata, but they have the exact same format. Labels tend to be the
+more common and represent a short name for a cue point. Notes might be used to represent a longer
+comment.
+*/
+typedef struct
+{
+ /* The ID of a cue point that this label or note corresponds to. */
+ drwav_uint32 cuePointId;
+
+ /* Size of the string not including any null terminator. */
+ drwav_uint32 stringLength;
+
+ /* The string. The *init_with_metadata functions null terminate this for convenience. */
+ char* pString;
+} drwav_list_label_or_note;
+
+/*
+BEXT metadata, also known as Broadcast Wave Format (BWF)
+
+This metadata adds some extra description to an audio file. You must check the version field to
+determine if the UMID or the loudness fields are valid.
+*/
+typedef struct
+{
+ /*
+ These top 3 fields, and the umid field are actually defined in the standard as a statically
+ sized buffers. In order to reduce the size of this struct (and therefore the union in the
+ metadata struct), we instead store these as pointers.
+ */
+ char* pDescription; /* Can be NULL or a null-terminated string, must be <= 256 characters. */
+ char* pOriginatorName; /* Can be NULL or a null-terminated string, must be <= 32 characters. */
+ char* pOriginatorReference; /* Can be NULL or a null-terminated string, must be <= 32 characters. */
+ char pOriginationDate[10]; /* ASCII "yyyy:mm:dd". */
+ char pOriginationTime[8]; /* ASCII "hh:mm:ss". */
+ drwav_uint64 timeReference; /* First sample count since midnight. */
+ drwav_uint16 version; /* Version of the BWF, check this to see if the fields below are valid. */
+
+ /*
+ Unrestricted ASCII characters containing a collection of strings terminated by CR/LF. Each
+ string shall contain a description of a coding process applied to the audio data.
+ */
+ char* pCodingHistory;
+ drwav_uint32 codingHistorySize;
+
+ /* Fields below this point are only valid if the version is 1 or above. */
+ drwav_uint8* pUMID; /* Exactly 64 bytes of SMPTE UMID */
+
+ /* Fields below this point are only valid if the version is 2 or above. */
+ drwav_uint16 loudnessValue; /* Integrated Loudness Value of the file in LUFS (multiplied by 100). */
+ drwav_uint16 loudnessRange; /* Loudness Range of the file in LU (multiplied by 100). */
+ drwav_uint16 maxTruePeakLevel; /* Maximum True Peak Level of the file expressed as dBTP (multiplied by 100). */
+ drwav_uint16 maxMomentaryLoudness; /* Highest value of the Momentary Loudness Level of the file in LUFS (multiplied by 100). */
+ drwav_uint16 maxShortTermLoudness; /* Highest value of the Short-Term Loudness Level of the file in LUFS (multiplied by 100). */
+} drwav_bext;
+
+/*
+Info Text Metadata
+
+There a many different types of information text that can be saved in this format. This is where
+things like the album name, the artists, the year it was produced, etc are saved. See
+drwav_metadata_type for the full list of types that dr_wav supports.
+*/
+typedef struct
+{
+ /* Size of the string not including any null terminator. */
+ drwav_uint32 stringLength;
+
+ /* The string. The *init_with_metadata functions null terminate this for convenience. */
+ char* pString;
+} drwav_list_info_text;
+
+/*
+Labelled Cue Region Metadata
+
+The labelled cue region metadata is used to associate some region of audio with text. The region
+starts at a cue point, and extends for the given number of samples.
+*/
+typedef struct
+{
+ /* The ID of a cue point that this object corresponds to. */
+ drwav_uint32 cuePointId;
+
+ /* The number of samples from the cue point forwards that should be considered this region */
+ drwav_uint32 sampleLength;
+
+ /* Four characters used to say what the purpose of this region is. */
+ drwav_uint8 purposeId[4];
+
+ /* Unsure of the exact meanings of these. It appears to be acceptable to set them all to 0. */
+ drwav_uint16 country;
+ drwav_uint16 language;
+ drwav_uint16 dialect;
+ drwav_uint16 codePage;
+
+ /* Size of the string not including any null terminator. */
+ drwav_uint32 stringLength;
+
+ /* The string. The *init_with_metadata functions null terminate this for convenience. */
+ char* pString;
+} drwav_list_labelled_cue_region;
+
+/*
+Unknown Metadata
+
+This chunk just represents a type of chunk that dr_wav does not understand.
+
+Unknown metadata has a location attached to it. This is because wav files can have a LIST chunk
+that contains subchunks. These LIST chunks can be one of two types. An adtl list, or an INFO
+list. This enum is used to specify the location of a chunk that dr_wav currently doesn't support.
+*/
+typedef enum
+{
+ drwav_metadata_location_invalid,
+ drwav_metadata_location_top_level,
+ drwav_metadata_location_inside_info_list,
+ drwav_metadata_location_inside_adtl_list
+} drwav_metadata_location;
+
+typedef struct
+{
+ drwav_uint8 id[4];
+ drwav_metadata_location chunkLocation;
+ drwav_uint32 dataSizeInBytes;
+ drwav_uint8* pData;
+} drwav_unknown_metadata;
+
+/*
+Metadata is saved as a union of all the supported types.
+*/
+typedef struct
+{
+ /* Determines which item in the union is valid. */
+ drwav_metadata_type type;
+
+ union
+ {
+ drwav_cue cue;
+ drwav_smpl smpl;
+ drwav_acid acid;
+ drwav_inst inst;
+ drwav_bext bext;
+ drwav_list_label_or_note labelOrNote; /* List label or list note. */
+ drwav_list_labelled_cue_region labelledCueRegion;
+ drwav_list_info_text infoText; /* Any of the list info types. */
+ drwav_unknown_metadata unknown;
+ } data;
+} drwav_metadata;
+
+typedef struct
+{
+ /* A pointer to the function to call when more data is needed. */
+ drwav_read_proc onRead;
+
+ /* A pointer to the function to call when data needs to be written. Only used when the drwav object is opened in write mode. */
+ drwav_write_proc onWrite;
+
+ /* A pointer to the function to call when the wav file needs to be seeked. */
+ drwav_seek_proc onSeek;
+
+ /* A pointer to the function to call when the position of the stream needs to be retrieved. */
+ drwav_tell_proc onTell;
+
+ /* The user data to pass to callbacks. */
+ void* pUserData;
+
+ /* Allocation callbacks. */
+ drwav_allocation_callbacks allocationCallbacks;
+
+
+ /* Whether or not the WAV file is formatted as a standard RIFF file or W64. */
+ drwav_container container;
+
+
+ /* Structure containing format information exactly as specified by the wav file. */
+ drwav_fmt fmt;
+
+ /* The sample rate. Will be set to something like 44100. */
+ drwav_uint32 sampleRate;
+
+ /* The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. */
+ drwav_uint16 channels;
+
+ /* The bits per sample. Will be set to something like 16, 24, etc. */
+ drwav_uint16 bitsPerSample;
+
+ /* Equal to fmt.formatTag, or the value specified by fmt.subFormat if fmt.formatTag is equal to 65534 (WAVE_FORMAT_EXTENSIBLE). */
+ drwav_uint16 translatedFormatTag;
+
+ /* The total number of PCM frames making up the audio data. */
+ drwav_uint64 totalPCMFrameCount;
+
+
+ /* The size in bytes of the data chunk. */
+ drwav_uint64 dataChunkDataSize;
+
+ /* The position in the stream of the first data byte of the data chunk. This is used for seeking. */
+ drwav_uint64 dataChunkDataPos;
+
+ /* The number of bytes remaining in the data chunk. */
+ drwav_uint64 bytesRemaining;
+
+ /* The current read position in PCM frames. */
+ drwav_uint64 readCursorInPCMFrames;
+
+
+ /*
+ Only used in sequential write mode. Keeps track of the desired size of the "data" chunk at the point of initialization time. Always
+ set to 0 for non-sequential writes and when the drwav object is opened in read mode. Used for validation.
+ */
+ drwav_uint64 dataChunkDataSizeTargetWrite;
+
+ /* Keeps track of whether or not the wav writer was initialized in sequential mode. */
+ drwav_bool32 isSequentialWrite;
+
+
+ /* A array of metadata. This is valid after the *init_with_metadata call returns. It will be valid until drwav_uninit() is called. You can take ownership of this data with drwav_take_ownership_of_metadata(). */
+ drwav_metadata* pMetadata;
+ drwav_uint32 metadataCount;
+
+
+ /* A hack to avoid a DRWAV_MALLOC() when opening a decoder with drwav_init_memory(). */
+ drwav__memory_stream memoryStream;
+ drwav__memory_stream_write memoryStreamWrite;
+
+
+ /* Microsoft ADPCM specific data. */
+ struct
+ {
+ drwav_uint32 bytesRemainingInBlock;
+ drwav_uint16 predictor[2];
+ drwav_int32 delta[2];
+ drwav_int32 cachedFrames[4]; /* Samples are stored in this cache during decoding. */
+ drwav_uint32 cachedFrameCount;
+ drwav_int32 prevFrames[2][2]; /* The previous 2 samples for each channel (2 channels at most). */
+ } msadpcm;
+
+ /* IMA ADPCM specific data. */
+ struct
+ {
+ drwav_uint32 bytesRemainingInBlock;
+ drwav_int32 predictor[2];
+ drwav_int32 stepIndex[2];
+ drwav_int32 cachedFrames[16]; /* Samples are stored in this cache during decoding. */
+ drwav_uint32 cachedFrameCount;
+ } ima;
+
+ /* AIFF specific data. */
+ struct
+ {
+ drwav_bool8 isLE; /* Will be set to true if the audio data is little-endian encoded. */
+ drwav_bool8 isUnsigned; /* Only used for 8-bit samples. When set to true, will be treated as unsigned. */
+ } aiff;
+} drwav;
+
+
+/*
+Initializes a pre-allocated drwav object for reading.
+
+pWav [out] A pointer to the drwav object being initialized.
+onRead [in] The function to call when data needs to be read from the client.
+onSeek [in] The function to call when the read position of the client data needs to move.
+onChunk [in, optional] The function to call when a chunk is enumerated at initialized time.
+pUserData, pReadSeekUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek.
+pChunkUserData [in, optional] A pointer to application defined data that will be passed to onChunk.
+flags [in, optional] A set of flags for controlling how things are loaded.
+
+Returns true if successful; false otherwise.
+
+Close the loader with drwav_uninit().
+
+This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory()
+to open the stream from a file or from a block of memory respectively.
+
+Possible values for flags:
+ DRWAV_SEQUENTIAL: Never perform a backwards seek while loading. This disables the chunk callback and will cause this function
+ to return as soon as the data chunk is found. Any chunks after the data chunk will be ignored.
+
+drwav_init() is equivalent to "drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0);".
+
+The onChunk callback is not called for the WAVE or FMT chunks. The contents of the FMT chunk can be read from pWav->fmt
+after the function returns.
+
+See also: drwav_init_file(), drwav_init_memory(), drwav_uninit()
+*/
+DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, drwav_chunk_proc onChunk, void* pReadSeekTellUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+
+/*
+Initializes a pre-allocated drwav object for writing.
+
+onWrite [in] The function to call when data needs to be written.
+onSeek [in] The function to call when the write position needs to move.
+pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek.
+metadata, numMetadata [in, optional] An array of metadata objects that should be written to the file. The array is not edited. You are responsible for this metadata memory and it must maintain valid until drwav_uninit() is called.
+
+Returns true if successful; false otherwise.
+
+Close the writer with drwav_uninit().
+
+This is the lowest level function for initializing a WAV file. You can also use drwav_init_file_write() and drwav_init_memory_write()
+to open the stream from a file or from a block of memory respectively.
+
+If the total sample count is known, you can use drwav_init_write_sequential(). This avoids the need for dr_wav to perform
+a post-processing step for storing the total sample count and the size of the data chunk which requires a backwards seek.
+
+See also: drwav_init_file_write(), drwav_init_memory_write(), drwav_uninit()
+*/
+DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
+
+/*
+Utility function to determine the target size of the entire data to be written (including all headers and chunks).
+
+Returns the target size in bytes.
+
+The metadata argument can be NULL meaning no metadata exists.
+
+Useful if the application needs to know the size to allocate.
+
+Only writing to the RIFF chunk and one data chunk is currently supported.
+
+See also: drwav_init_write(), drwav_init_file_write(), drwav_init_memory_write()
+*/
+DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
+
+/*
+Take ownership of the metadata objects that were allocated via one of the init_with_metadata() function calls. The init_with_metdata functions perform a single heap allocation for this metadata.
+
+Useful if you want the data to persist beyond the lifetime of the drwav object.
+
+You must free the data returned from this function using drwav_free().
+*/
+DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav);
+
+/*
+Uninitializes the given drwav object.
+
+Use this only for objects initialized with drwav_init*() functions (drwav_init(), drwav_init_ex(), drwav_init_write(), drwav_init_write_sequential()).
+*/
+DRWAV_API drwav_result drwav_uninit(drwav* pWav);
+
+
+/*
+Reads raw audio data.
+
+This is the lowest level function for reading audio data. It simply reads the given number of
+bytes of the raw internal sample data.
+
+Consider using drwav_read_pcm_frames_s16(), drwav_read_pcm_frames_s32() or drwav_read_pcm_frames_f32() for
+reading sample data in a consistent format.
+
+pBufferOut can be NULL in which case a seek will be performed.
+
+Returns the number of bytes actually read.
+*/
+DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut);
+
+/*
+Reads up to the specified number of PCM frames from the WAV file.
+
+The output data will be in the file's internal format, converted to native-endian byte order. Use
+drwav_read_pcm_frames_s16/f32/s32() to read data in a specific format.
+
+If the return value is less than <framesToRead> it means the end of the file has been reached or
+you have requested more PCM frames than can possibly fit in the output buffer.
+
+This function will only work when sample data is of a fixed size and uncompressed. If you are
+using a compressed format consider using drwav_read_raw() or drwav_read_pcm_frames_s16/s32/f32().
+
+pBufferOut can be NULL in which case a seek will be performed.
+*/
+DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
+
+/*
+Seeks to the given PCM frame.
+
+Returns true if successful; false otherwise.
+*/
+DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex);
+
+/*
+Retrieves the current read position in pcm frames.
+*/
+DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor);
+
+/*
+Retrieves the length of the file.
+*/
+DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength);
+
+
+/*
+Writes raw audio data.
+
+Returns the number of bytes actually written. If this differs from bytesToWrite, it indicates an error.
+*/
+DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData);
+
+/*
+Writes PCM frames.
+
+Returns the number of PCM frames written.
+
+Input samples need to be in native-endian byte order. On big-endian architectures the input data will be converted to
+little-endian. Use drwav_write_raw() to write raw audio data without performing any conversion.
+*/
+DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
+DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
+DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
+
+/* Conversion Utilities */
+#ifndef DR_WAV_NO_CONVERSION_API
+
+/*
+Reads a chunk of audio data and converts it to signed 16-bit PCM samples.
+
+pBufferOut can be NULL in which case a seek will be performed.
+
+Returns the number of PCM frames actually read.
+
+If the return value is less than <framesToRead> it means the end of the file has been reached.
+*/
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
+
+/* Low-level function for converting unsigned 8-bit PCM samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 24-bit PCM samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 32-bit PCM samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount);
+
+/* Low-level function for converting IEEE 32-bit floating point samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount);
+
+/* Low-level function for converting IEEE 64-bit floating point samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount);
+
+/* Low-level function for converting A-law samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting u-law samples to signed 16-bit PCM samples. */
+DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+
+/*
+Reads a chunk of audio data and converts it to IEEE 32-bit floating point samples.
+
+pBufferOut can be NULL in which case a seek will be performed.
+
+Returns the number of PCM frames actually read.
+
+If the return value is less than <framesToRead> it means the end of the file has been reached.
+*/
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
+
+/* Low-level function for converting unsigned 8-bit PCM samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 16-bit PCM samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 24-bit PCM samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 32-bit PCM samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount);
+
+/* Low-level function for converting IEEE 64-bit floating point samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount);
+
+/* Low-level function for converting A-law samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting u-law samples to IEEE 32-bit floating point samples. */
+DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+
+/*
+Reads a chunk of audio data and converts it to signed 32-bit PCM samples.
+
+pBufferOut can be NULL in which case a seek will be performed.
+
+Returns the number of PCM frames actually read.
+
+If the return value is less than <framesToRead> it means the end of the file has been reached.
+*/
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
+
+/* Low-level function for converting unsigned 8-bit PCM samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 16-bit PCM samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount);
+
+/* Low-level function for converting signed 24-bit PCM samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting IEEE 32-bit floating point samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount);
+
+/* Low-level function for converting IEEE 64-bit floating point samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount);
+
+/* Low-level function for converting A-law samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+/* Low-level function for converting u-law samples to signed 32-bit PCM samples. */
+DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+
+#endif /* DR_WAV_NO_CONVERSION_API */
+
+
+/* High-Level Convenience Helpers */
+
+#ifndef DR_WAV_NO_STDIO
+/*
+Helper for initializing a wave file for reading using stdio.
+
+This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav
+objects because the operating system may restrict the number of file handles an application can have open at
+any given time.
+*/
+DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Helper for initializing a wave file for writing using stdio.
+
+This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav
+objects because the operating system may restrict the number of file handles an application can have open at
+any given time.
+*/
+DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+#endif /* DR_WAV_NO_STDIO */
+
+/*
+Helper for initializing a loader from a pre-allocated memory buffer.
+
+This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for
+the lifetime of the drwav object.
+
+The buffer should contain the contents of the entire wave file, not just the sample data.
+*/
+DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+
+/*
+Helper for initializing a writer which outputs data to a memory buffer.
+
+dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free().
+
+The buffer will remain allocated even after drwav_uninit() is called. The buffer should not be considered valid
+until after drwav_uninit() has been called.
+*/
+DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+
+
+#ifndef DR_WAV_NO_CONVERSION_API
+/*
+Opens and reads an entire wav file in a single operation.
+
+The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer.
+*/
+DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+#ifndef DR_WAV_NO_STDIO
+/*
+Opens and decodes an entire wav file in a single operation.
+
+The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer.
+*/
+DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+#endif
+/*
+Opens and decodes an entire wav file from a block of memory in a single operation.
+
+The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer.
+*/
+DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+#endif
+
+/* Frees data that was allocated internally by dr_wav. */
+DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks);
+
+/* Converts bytes from a wav stream to a sized type of native endian. */
+DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data);
+DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data);
+DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data);
+DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data);
+DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data);
+DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data);
+DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data);
+
+/* Compares a GUID for the purpose of checking the type of a Wave64 chunk. */
+DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]);
+
+/* Compares a four-character-code for the purpose of checking the type of a RIFF chunk. */
+DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b);
+
+#ifdef __cplusplus
+}
+#endif
+#endif /* dr_wav_h */
+
+
+/************************************************************************************************************************************************************
+ ************************************************************************************************************************************************************
+
+ IMPLEMENTATION
+
+ ************************************************************************************************************************************************************
+ ************************************************************************************************************************************************************/
+#if 1
+#ifndef dr_wav_c
+#define dr_wav_c
+
+#ifdef __MRC__
+/* MrC currently doesn't compile dr_wav correctly with any optimizations enabled. */
+#pragma options opt off
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <limits.h> /* For INT_MAX */
+
+#ifndef DR_WAV_NO_STDIO
+#include <stdio.h>
+#ifndef DR_WAV_NO_WCHAR
+#include <wchar.h>
+#endif
+#endif
+
+/* Standard library stuff. */
+#ifndef DRWAV_ASSERT
+#include <assert.h>
+#define DRWAV_ASSERT(expression) assert(expression)
+#endif
+#ifndef DRWAV_MALLOC
+#define DRWAV_MALLOC(sz) malloc((sz))
+#endif
+#ifndef DRWAV_REALLOC
+#define DRWAV_REALLOC(p, sz) realloc((p), (sz))
+#endif
+#ifndef DRWAV_FREE
+#define DRWAV_FREE(p) free((p))
+#endif
+#ifndef DRWAV_COPY_MEMORY
+#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz))
+#endif
+#ifndef DRWAV_ZERO_MEMORY
+#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz))
+#endif
+#ifndef DRWAV_ZERO_OBJECT
+#define DRWAV_ZERO_OBJECT(p) DRWAV_ZERO_MEMORY((p), sizeof(*p))
+#endif
+
+#define drwav_countof(x) (sizeof(x) / sizeof(x[0]))
+#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a))
+#define drwav_min(a, b) (((a) < (b)) ? (a) : (b))
+#define drwav_max(a, b) (((a) > (b)) ? (a) : (b))
+#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x))))
+#define drwav_offset_ptr(p, offset) (((drwav_uint8*)(p)) + (offset))
+
+#define DRWAV_MAX_SIMD_VECTOR_SIZE 32
+
+/* Architecture Detection */
+#if defined(__x86_64__) || (defined(_M_X64) && !defined(_M_ARM64EC))
+ #define DRWAV_X64
+#elif defined(__i386) || defined(_M_IX86)
+ #define DRWAV_X86
+#elif defined(__arm__) || defined(_M_ARM)
+ #define DRWAV_ARM
+#endif
+/* End Architecture Detection */
+
+/* Inline */
+#ifdef _MSC_VER
+ #define DRWAV_INLINE __forceinline
+#elif defined(__GNUC__)
+ /*
+ I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when
+ the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some
+ case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the
+ command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue
+ I am using "__inline__" only when we're compiling in strict ANSI mode.
+ */
+ #if defined(__STRICT_ANSI__)
+ #define DRWAV_GNUC_INLINE_HINT __inline__
+ #else
+ #define DRWAV_GNUC_INLINE_HINT inline
+ #endif
+
+ #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__)
+ #define DRWAV_INLINE DRWAV_GNUC_INLINE_HINT __attribute__((always_inline))
+ #else
+ #define DRWAV_INLINE DRWAV_GNUC_INLINE_HINT
+ #endif
+#elif defined(__WATCOMC__)
+ #define DRWAV_INLINE __inline
+#else
+ #define DRWAV_INLINE
+#endif
+/* End Inline */
+
+/* SIZE_MAX */
+#if defined(SIZE_MAX)
+ #define DRWAV_SIZE_MAX SIZE_MAX
+#else
+ #if defined(_WIN64) || defined(_LP64) || defined(__LP64__)
+ #define DRWAV_SIZE_MAX ((drwav_uint64)0xFFFFFFFFFFFFFFFF)
+ #else
+ #define DRWAV_SIZE_MAX 0xFFFFFFFF
+ #endif
+#endif
+/* End SIZE_MAX */
+
+/* Weird bit manipulation is for C89 compatibility (no direct support for 64-bit integers). */
+#define DRWAV_INT64_MIN ((drwav_int64) ((drwav_uint64)0x80000000 << 32))
+#define DRWAV_INT64_MAX ((drwav_int64)(((drwav_uint64)0x7FFFFFFF << 32) | 0xFFFFFFFF))
+
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ #define DRWAV_HAS_BYTESWAP16_INTRINSIC
+ #define DRWAV_HAS_BYTESWAP32_INTRINSIC
+ #define DRWAV_HAS_BYTESWAP64_INTRINSIC
+#elif defined(__clang__)
+ #if defined(__has_builtin)
+ #if __has_builtin(__builtin_bswap16)
+ #define DRWAV_HAS_BYTESWAP16_INTRINSIC
+ #endif
+ #if __has_builtin(__builtin_bswap32)
+ #define DRWAV_HAS_BYTESWAP32_INTRINSIC
+ #endif
+ #if __has_builtin(__builtin_bswap64)
+ #define DRWAV_HAS_BYTESWAP64_INTRINSIC
+ #endif
+ #endif
+#elif defined(__GNUC__)
+ #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3))
+ #define DRWAV_HAS_BYTESWAP32_INTRINSIC
+ #define DRWAV_HAS_BYTESWAP64_INTRINSIC
+ #endif
+ #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
+ #define DRWAV_HAS_BYTESWAP16_INTRINSIC
+ #endif
+#endif
+
+DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision)
+{
+ if (pMajor) {
+ *pMajor = DRWAV_VERSION_MAJOR;
+ }
+
+ if (pMinor) {
+ *pMinor = DRWAV_VERSION_MINOR;
+ }
+
+ if (pRevision) {
+ *pRevision = DRWAV_VERSION_REVISION;
+ }
+}
+
+DRWAV_API const char* drwav_version_string(void)
+{
+ return DRWAV_VERSION_STRING;
+}
+
+/*
+These limits are used for basic validation when initializing the decoder. If you exceed these limits, first of all: what on Earth are
+you doing?! (Let me know, I'd be curious!) Second, you can adjust these by #define-ing them before the dr_wav implementation.
+*/
+#ifndef DRWAV_MAX_SAMPLE_RATE
+#define DRWAV_MAX_SAMPLE_RATE 384000
+#endif
+#ifndef DRWAV_MAX_CHANNELS
+#define DRWAV_MAX_CHANNELS 256
+#endif
+#ifndef DRWAV_MAX_BITS_PER_SAMPLE
+#define DRWAV_MAX_BITS_PER_SAMPLE 64
+#endif
+
+static const drwav_uint8 drwavGUID_W64_RIFF[16] = {0x72,0x69,0x66,0x66, 0x2E,0x91, 0xCF,0x11, 0xA5,0xD6, 0x28,0xDB,0x04,0xC1,0x00,0x00}; /* 66666972-912E-11CF-A5D6-28DB04C10000 */
+static const drwav_uint8 drwavGUID_W64_WAVE[16] = {0x77,0x61,0x76,0x65, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 65766177-ACF3-11D3-8CD1-00C04F8EDB8A */
+/*static const drwav_uint8 drwavGUID_W64_JUNK[16] = {0x6A,0x75,0x6E,0x6B, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6B6E756A-ACF3-11D3-8CD1-00C04F8EDB8A */
+static const drwav_uint8 drwavGUID_W64_FMT [16] = {0x66,0x6D,0x74,0x20, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 20746D66-ACF3-11D3-8CD1-00C04F8EDB8A */
+static const drwav_uint8 drwavGUID_W64_FACT[16] = {0x66,0x61,0x63,0x74, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 74636166-ACF3-11D3-8CD1-00C04F8EDB8A */
+static const drwav_uint8 drwavGUID_W64_DATA[16] = {0x64,0x61,0x74,0x61, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 61746164-ACF3-11D3-8CD1-00C04F8EDB8A */
+/*static const drwav_uint8 drwavGUID_W64_SMPL[16] = {0x73,0x6D,0x70,0x6C, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6C706D73-ACF3-11D3-8CD1-00C04F8EDB8A */
+
+
+static DRWAV_INLINE int drwav__is_little_endian(void)
+{
+#if defined(DRWAV_X86) || defined(DRWAV_X64)
+ return DRWAV_TRUE;
+#elif defined(__BYTE_ORDER) && defined(__LITTLE_ENDIAN) && __BYTE_ORDER == __LITTLE_ENDIAN
+ return DRWAV_TRUE;
+#else
+ int n = 1;
+ return (*(char*)&n) == 1;
+#endif
+}
+
+
+static DRWAV_INLINE void drwav_bytes_to_guid(const drwav_uint8* data, drwav_uint8* guid)
+{
+ int i;
+ for (i = 0; i < 16; ++i) {
+ guid[i] = data[i];
+ }
+}
+
+
+static DRWAV_INLINE drwav_uint16 drwav__bswap16(drwav_uint16 n)
+{
+#ifdef DRWAV_HAS_BYTESWAP16_INTRINSIC
+ #if defined(_MSC_VER)
+ return _byteswap_ushort(n);
+ #elif defined(__GNUC__) || defined(__clang__)
+ return __builtin_bswap16(n);
+ #else
+ #error "This compiler does not support the byte swap intrinsic."
+ #endif
+#else
+ return ((n & 0xFF00) >> 8) |
+ ((n & 0x00FF) << 8);
+#endif
+}
+
+static DRWAV_INLINE drwav_uint32 drwav__bswap32(drwav_uint32 n)
+{
+#ifdef DRWAV_HAS_BYTESWAP32_INTRINSIC
+ #if defined(_MSC_VER)
+ return _byteswap_ulong(n);
+ #elif defined(__GNUC__) || defined(__clang__)
+ #if defined(DRWAV_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(DRWAV_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */
+ /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */
+ drwav_uint32 r;
+ __asm__ __volatile__ (
+ #if defined(DRWAV_64BIT)
+ "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */
+ #else
+ "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n)
+ #endif
+ );
+ return r;
+ #else
+ return __builtin_bswap32(n);
+ #endif
+ #else
+ #error "This compiler does not support the byte swap intrinsic."
+ #endif
+#else
+ return ((n & 0xFF000000) >> 24) |
+ ((n & 0x00FF0000) >> 8) |
+ ((n & 0x0000FF00) << 8) |
+ ((n & 0x000000FF) << 24);
+#endif
+}
+
+static DRWAV_INLINE drwav_uint64 drwav__bswap64(drwav_uint64 n)
+{
+#ifdef DRWAV_HAS_BYTESWAP64_INTRINSIC
+ #if defined(_MSC_VER)
+ return _byteswap_uint64(n);
+ #elif defined(__GNUC__) || defined(__clang__)
+ return __builtin_bswap64(n);
+ #else
+ #error "This compiler does not support the byte swap intrinsic."
+ #endif
+#else
+ /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */
+ return ((n & ((drwav_uint64)0xFF000000 << 32)) >> 56) |
+ ((n & ((drwav_uint64)0x00FF0000 << 32)) >> 40) |
+ ((n & ((drwav_uint64)0x0000FF00 << 32)) >> 24) |
+ ((n & ((drwav_uint64)0x000000FF << 32)) >> 8) |
+ ((n & ((drwav_uint64)0xFF000000 )) << 8) |
+ ((n & ((drwav_uint64)0x00FF0000 )) << 24) |
+ ((n & ((drwav_uint64)0x0000FF00 )) << 40) |
+ ((n & ((drwav_uint64)0x000000FF )) << 56);
+#endif
+}
+
+
+static DRWAV_INLINE drwav_int16 drwav__bswap_s16(drwav_int16 n)
+{
+ return (drwav_int16)drwav__bswap16((drwav_uint16)n);
+}
+
+static DRWAV_INLINE void drwav__bswap_samples_s16(drwav_int16* pSamples, drwav_uint64 sampleCount)
+{
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamples[iSample] = drwav__bswap_s16(pSamples[iSample]);
+ }
+}
+
+
+static DRWAV_INLINE void drwav__bswap_s24(drwav_uint8* p)
+{
+ drwav_uint8 t;
+ t = p[0];
+ p[0] = p[2];
+ p[2] = t;
+}
+
+static DRWAV_INLINE void drwav__bswap_samples_s24(drwav_uint8* pSamples, drwav_uint64 sampleCount)
+{
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ drwav_uint8* pSample = pSamples + (iSample*3);
+ drwav__bswap_s24(pSample);
+ }
+}
+
+
+static DRWAV_INLINE drwav_int32 drwav__bswap_s32(drwav_int32 n)
+{
+ return (drwav_int32)drwav__bswap32((drwav_uint32)n);
+}
+
+static DRWAV_INLINE void drwav__bswap_samples_s32(drwav_int32* pSamples, drwav_uint64 sampleCount)
+{
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamples[iSample] = drwav__bswap_s32(pSamples[iSample]);
+ }
+}
+
+
+static DRWAV_INLINE drwav_int64 drwav__bswap_s64(drwav_int64 n)
+{
+ return (drwav_int64)drwav__bswap64((drwav_uint64)n);
+}
+
+static DRWAV_INLINE void drwav__bswap_samples_s64(drwav_int64* pSamples, drwav_uint64 sampleCount)
+{
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamples[iSample] = drwav__bswap_s64(pSamples[iSample]);
+ }
+}
+
+
+static DRWAV_INLINE float drwav__bswap_f32(float n)
+{
+ union {
+ drwav_uint32 i;
+ float f;
+ } x;
+ x.f = n;
+ x.i = drwav__bswap32(x.i);
+
+ return x.f;
+}
+
+static DRWAV_INLINE void drwav__bswap_samples_f32(float* pSamples, drwav_uint64 sampleCount)
+{
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamples[iSample] = drwav__bswap_f32(pSamples[iSample]);
+ }
+}
+
+
+static DRWAV_INLINE void drwav__bswap_samples(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample)
+{
+ switch (bytesPerSample)
+ {
+ case 1:
+ {
+ /* No-op. */
+ } break;
+ case 2:
+ {
+ drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount);
+ } break;
+ case 3:
+ {
+ drwav__bswap_samples_s24((drwav_uint8*)pSamples, sampleCount);
+ } break;
+ case 4:
+ {
+ drwav__bswap_samples_s32((drwav_int32*)pSamples, sampleCount);
+ } break;
+ case 8:
+ {
+ drwav__bswap_samples_s64((drwav_int64*)pSamples, sampleCount);
+ } break;
+ default:
+ {
+ /* Unsupported format. */
+ DRWAV_ASSERT(DRWAV_FALSE);
+ } break;
+ }
+}
+
+
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_bool32 drwav_is_container_be(drwav_container container)
+{
+ if (container == drwav_container_rifx || container == drwav_container_aiff) {
+ return DRWAV_TRUE;
+ } else {
+ return DRWAV_FALSE;
+ }
+}
+
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_uint16 drwav_bytes_to_u16_le(const drwav_uint8* data)
+{
+ return ((drwav_uint16)data[0] << 0) | ((drwav_uint16)data[1] << 8);
+}
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_uint16 drwav_bytes_to_u16_be(const drwav_uint8* data)
+{
+ return ((drwav_uint16)data[1] << 0) | ((drwav_uint16)data[0] << 8);
+}
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_uint16 drwav_bytes_to_u16_ex(const drwav_uint8* data, drwav_container container)
+{
+ if (drwav_is_container_be(container)) {
+ return drwav_bytes_to_u16_be(data);
+ } else {
+ return drwav_bytes_to_u16_le(data);
+ }
+}
+
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_uint32 drwav_bytes_to_u32_le(const drwav_uint8* data)
+{
+ return ((drwav_uint32)data[0] << 0) | ((drwav_uint32)data[1] << 8) | ((drwav_uint32)data[2] << 16) | ((drwav_uint32)data[3] << 24);
+}
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_uint32 drwav_bytes_to_u32_be(const drwav_uint8* data)
+{
+ return ((drwav_uint32)data[3] << 0) | ((drwav_uint32)data[2] << 8) | ((drwav_uint32)data[1] << 16) | ((drwav_uint32)data[0] << 24);
+}
+
+DRWAV_PRIVATE DRWAV_INLINE drwav_uint32 drwav_bytes_to_u32_ex(const drwav_uint8* data, drwav_container container)
+{
+ if (drwav_is_container_be(container)) {
+ return drwav_bytes_to_u32_be(data);
+ } else {
+ return drwav_bytes_to_u32_le(data);
+ }
+}
+
+
+
+DRWAV_PRIVATE drwav_int64 drwav_aiff_extented_to_s64(const drwav_uint8* data)
+{
+ drwav_uint32 exponent = ((drwav_uint32)data[0] << 8) | data[1];
+ drwav_uint64 hi = ((drwav_uint64)data[2] << 24) | ((drwav_uint64)data[3] << 16) | ((drwav_uint64)data[4] << 8) | ((drwav_uint64)data[5] << 0);
+ drwav_uint64 lo = ((drwav_uint64)data[6] << 24) | ((drwav_uint64)data[7] << 16) | ((drwav_uint64)data[8] << 8) | ((drwav_uint64)data[9] << 0);
+ drwav_uint64 significand = (hi << 32) | lo;
+ int sign = exponent >> 15;
+
+ /* Remove sign bit. */
+ exponent &= 0x7FFF;
+
+ /* Special cases. */
+ if (exponent == 0 && significand == 0) {
+ return 0;
+ } else if (exponent == 0x7FFF) {
+ return sign ? DRWAV_INT64_MIN : DRWAV_INT64_MAX; /* Infinite. */
+ }
+
+ exponent -= 16383;
+
+ if (exponent > 63) {
+ return sign ? DRWAV_INT64_MIN : DRWAV_INT64_MAX; /* Too big for a 64-bit integer. */
+ } else if (exponent < 1) {
+ return 0; /* Number is less than 1, so rounds down to 0. */
+ }
+
+ significand >>= (63 - exponent);
+
+ if (sign) {
+ return -(drwav_int64)significand;
+ } else {
+ return (drwav_int64)significand;
+ }
+}
+
+
+DRWAV_PRIVATE void* drwav__malloc_default(size_t sz, void* pUserData)
+{
+ (void)pUserData;
+ return DRWAV_MALLOC(sz);
+}
+
+DRWAV_PRIVATE void* drwav__realloc_default(void* p, size_t sz, void* pUserData)
+{
+ (void)pUserData;
+ return DRWAV_REALLOC(p, sz);
+}
+
+DRWAV_PRIVATE void drwav__free_default(void* p, void* pUserData)
+{
+ (void)pUserData;
+ DRWAV_FREE(p);
+}
+
+
+DRWAV_PRIVATE void* drwav__malloc_from_callbacks(size_t sz, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pAllocationCallbacks == NULL) {
+ return NULL;
+ }
+
+ if (pAllocationCallbacks->onMalloc != NULL) {
+ return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData);
+ }
+
+ /* Try using realloc(). */
+ if (pAllocationCallbacks->onRealloc != NULL) {
+ return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData);
+ }
+
+ return NULL;
+}
+
+DRWAV_PRIVATE void* drwav__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pAllocationCallbacks == NULL) {
+ return NULL;
+ }
+
+ if (pAllocationCallbacks->onRealloc != NULL) {
+ return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData);
+ }
+
+ /* Try emulating realloc() in terms of malloc()/free(). */
+ if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) {
+ void* p2;
+
+ p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData);
+ if (p2 == NULL) {
+ return NULL;
+ }
+
+ if (p != NULL) {
+ DRWAV_COPY_MEMORY(p2, p, szOld);
+ pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
+ }
+
+ return p2;
+ }
+
+ return NULL;
+}
+
+DRWAV_PRIVATE void drwav__free_from_callbacks(void* p, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (p == NULL || pAllocationCallbacks == NULL) {
+ return;
+ }
+
+ if (pAllocationCallbacks->onFree != NULL) {
+ pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
+ }
+}
+
+
+DRWAV_PRIVATE drwav_allocation_callbacks drwav_copy_allocation_callbacks_or_defaults(const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pAllocationCallbacks != NULL) {
+ /* Copy. */
+ return *pAllocationCallbacks;
+ } else {
+ /* Defaults. */
+ drwav_allocation_callbacks allocationCallbacks;
+ allocationCallbacks.pUserData = NULL;
+ allocationCallbacks.onMalloc = drwav__malloc_default;
+ allocationCallbacks.onRealloc = drwav__realloc_default;
+ allocationCallbacks.onFree = drwav__free_default;
+ return allocationCallbacks;
+ }
+}
+
+
+static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag)
+{
+ return
+ formatTag == DR_WAVE_FORMAT_ADPCM ||
+ formatTag == DR_WAVE_FORMAT_DVI_ADPCM;
+}
+
+DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_riff(drwav_uint64 chunkSize)
+{
+ return (unsigned int)(chunkSize % 2);
+}
+
+DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_w64(drwav_uint64 chunkSize)
+{
+ return (unsigned int)(chunkSize % 8);
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut);
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut);
+DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount);
+
+DRWAV_PRIVATE drwav_result drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_chunk_header* pHeaderOut)
+{
+ if (container == drwav_container_riff || container == drwav_container_rifx || container == drwav_container_rf64 || container == drwav_container_aiff) {
+ drwav_uint8 sizeInBytes[4];
+
+ if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) {
+ return DRWAV_AT_END;
+ }
+
+ if (onRead(pUserData, sizeInBytes, 4) != 4) {
+ return DRWAV_INVALID_FILE;
+ }
+
+ pHeaderOut->sizeInBytes = drwav_bytes_to_u32_ex(sizeInBytes, container);
+ pHeaderOut->paddingSize = drwav__chunk_padding_size_riff(pHeaderOut->sizeInBytes);
+
+ *pRunningBytesReadOut += 8;
+ } else if (container == drwav_container_w64) {
+ drwav_uint8 sizeInBytes[8];
+
+ if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) {
+ return DRWAV_AT_END;
+ }
+
+ if (onRead(pUserData, sizeInBytes, 8) != 8) {
+ return DRWAV_INVALID_FILE;
+ }
+
+ pHeaderOut->sizeInBytes = drwav_bytes_to_u64(sizeInBytes) - 24; /* <-- Subtract 24 because w64 includes the size of the header. */
+ pHeaderOut->paddingSize = drwav__chunk_padding_size_w64(pHeaderOut->sizeInBytes);
+ *pRunningBytesReadOut += 24;
+ } else {
+ return DRWAV_INVALID_FILE;
+ }
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData)
+{
+ drwav_uint64 bytesRemainingToSeek = offset;
+ while (bytesRemainingToSeek > 0) {
+ if (bytesRemainingToSeek > 0x7FFFFFFF) {
+ if (!onSeek(pUserData, 0x7FFFFFFF, DRWAV_SEEK_CUR)) {
+ return DRWAV_FALSE;
+ }
+ bytesRemainingToSeek -= 0x7FFFFFFF;
+ } else {
+ if (!onSeek(pUserData, (int)bytesRemainingToSeek, DRWAV_SEEK_CUR)) {
+ return DRWAV_FALSE;
+ }
+ bytesRemainingToSeek = 0;
+ }
+ }
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__seek_from_start(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData)
+{
+ if (offset <= 0x7FFFFFFF) {
+ return onSeek(pUserData, (int)offset, DRWAV_SEEK_SET);
+ }
+
+ /* Larger than 32-bit seek. */
+ if (!onSeek(pUserData, 0x7FFFFFFF, DRWAV_SEEK_SET)) {
+ return DRWAV_FALSE;
+ }
+ offset -= 0x7FFFFFFF;
+
+ for (;;) {
+ if (offset <= 0x7FFFFFFF) {
+ return onSeek(pUserData, (int)offset, DRWAV_SEEK_CUR);
+ }
+
+ if (!onSeek(pUserData, 0x7FFFFFFF, DRWAV_SEEK_CUR)) {
+ return DRWAV_FALSE;
+ }
+ offset -= 0x7FFFFFFF;
+ }
+
+ /* Should never get here. */
+ /*return DRWAV_TRUE; */
+}
+
+
+
+DRWAV_PRIVATE size_t drwav__on_read(drwav_read_proc onRead, void* pUserData, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor)
+{
+ size_t bytesRead;
+
+ DRWAV_ASSERT(onRead != NULL);
+ DRWAV_ASSERT(pCursor != NULL);
+
+ bytesRead = onRead(pUserData, pBufferOut, bytesToRead);
+ *pCursor += bytesRead;
+ return bytesRead;
+}
+
+#if 0
+DRWAV_PRIVATE drwav_bool32 drwav__on_seek(drwav_seek_proc onSeek, void* pUserData, int offset, drwav_seek_origin origin, drwav_uint64* pCursor)
+{
+ DRWAV_ASSERT(onSeek != NULL);
+ DRWAV_ASSERT(pCursor != NULL);
+
+ if (!onSeek(pUserData, offset, origin)) {
+ return DRWAV_FALSE;
+ }
+
+ if (origin == DRWAV_SEEK_SET) {
+ *pCursor = offset;
+ } else {
+ *pCursor += offset;
+ }
+
+ return DRWAV_TRUE;
+}
+#endif
+
+
+#define DRWAV_SMPL_BYTES 36
+#define DRWAV_SMPL_LOOP_BYTES 24
+#define DRWAV_INST_BYTES 7
+#define DRWAV_ACID_BYTES 24
+#define DRWAV_CUE_BYTES 4
+#define DRWAV_BEXT_BYTES 602
+#define DRWAV_BEXT_DESCRIPTION_BYTES 256
+#define DRWAV_BEXT_ORIGINATOR_NAME_BYTES 32
+#define DRWAV_BEXT_ORIGINATOR_REF_BYTES 32
+#define DRWAV_BEXT_RESERVED_BYTES 180
+#define DRWAV_BEXT_UMID_BYTES 64
+#define DRWAV_CUE_POINT_BYTES 24
+#define DRWAV_LIST_LABEL_OR_NOTE_BYTES 4
+#define DRWAV_LIST_LABELLED_TEXT_BYTES 20
+
+#define DRWAV_METADATA_ALIGNMENT 8
+
+typedef enum
+{
+ drwav__metadata_parser_stage_count,
+ drwav__metadata_parser_stage_read
+} drwav__metadata_parser_stage;
+
+typedef struct
+{
+ drwav_read_proc onRead;
+ drwav_seek_proc onSeek;
+ void *pReadSeekUserData;
+ drwav__metadata_parser_stage stage;
+ drwav_metadata *pMetadata;
+ drwav_uint32 metadataCount;
+ drwav_uint8 *pData;
+ drwav_uint8 *pDataCursor;
+ drwav_uint64 metadataCursor;
+ drwav_uint64 extraCapacity;
+} drwav__metadata_parser;
+
+DRWAV_PRIVATE size_t drwav__metadata_memory_capacity(drwav__metadata_parser* pParser)
+{
+ drwav_uint64 cap = sizeof(drwav_metadata) * (drwav_uint64)pParser->metadataCount + pParser->extraCapacity;
+ if (cap > DRWAV_SIZE_MAX) {
+ return 0; /* Too big. */
+ }
+
+ return (size_t)cap; /* Safe cast thanks to the check above. */
+}
+
+DRWAV_PRIVATE drwav_uint8* drwav__metadata_get_memory(drwav__metadata_parser* pParser, size_t size, size_t align)
+{
+ drwav_uint8* pResult;
+
+ if (align) {
+ drwav_uintptr modulo = (drwav_uintptr)pParser->pDataCursor % align;
+ if (modulo != 0) {
+ pParser->pDataCursor += align - modulo;
+ }
+ }
+
+ pResult = pParser->pDataCursor;
+
+ /*
+ Getting to the point where this function is called means there should always be memory
+ available. Out of memory checks should have been done at an earlier stage.
+ */
+ DRWAV_ASSERT((pResult + size) <= (pParser->pData + drwav__metadata_memory_capacity(pParser)));
+
+ pParser->pDataCursor += size;
+ return pResult;
+}
+
+DRWAV_PRIVATE void drwav__metadata_request_extra_memory_for_stage_2(drwav__metadata_parser* pParser, size_t bytes, size_t align)
+{
+ size_t extra = bytes + (align ? (align - 1) : 0);
+ pParser->extraCapacity += extra;
+}
+
+DRWAV_PRIVATE drwav_result drwav__metadata_alloc(drwav__metadata_parser* pParser, drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pParser->extraCapacity != 0 || pParser->metadataCount != 0) {
+ pAllocationCallbacks->onFree(pParser->pData, pAllocationCallbacks->pUserData);
+
+ pParser->pData = (drwav_uint8*)pAllocationCallbacks->onMalloc(drwav__metadata_memory_capacity(pParser), pAllocationCallbacks->pUserData);
+ pParser->pDataCursor = pParser->pData;
+
+ if (pParser->pData == NULL) {
+ return DRWAV_OUT_OF_MEMORY;
+ }
+
+ /*
+ We don't need to worry about specifying an alignment here because malloc always returns something
+ of suitable alignment. This also means pParser->pMetadata is all that we need to store in order
+ for us to free when we are done.
+ */
+ pParser->pMetadata = (drwav_metadata*)drwav__metadata_get_memory(pParser, sizeof(drwav_metadata) * pParser->metadataCount, 1);
+ pParser->metadataCursor = 0;
+ }
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_PRIVATE size_t drwav__metadata_parser_read(drwav__metadata_parser* pParser, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor)
+{
+ if (pCursor != NULL) {
+ return drwav__on_read(pParser->onRead, pParser->pReadSeekUserData, pBufferOut, bytesToRead, pCursor);
+ } else {
+ return pParser->onRead(pParser->pReadSeekUserData, pBufferOut, bytesToRead);
+ }
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_smpl_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata)
+{
+ drwav_uint8 smplHeaderData[DRWAV_SMPL_BYTES];
+ drwav_uint64 totalBytesRead = 0;
+ size_t bytesJustRead;
+
+ if (pMetadata == NULL) {
+ return 0;
+ }
+
+ bytesJustRead = drwav__metadata_parser_read(pParser, smplHeaderData, sizeof(smplHeaderData), &totalBytesRead);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+ DRWAV_ASSERT(pChunkHeader != NULL);
+
+ if (pMetadata != NULL && bytesJustRead == sizeof(smplHeaderData)) {
+ drwav_uint32 iSampleLoop;
+ drwav_uint32 loopCount;
+ drwav_uint32 calculatedLoopCount;
+
+ /*
+ When we calcualted the amount of memory required for the "smpl" chunk we excluded the chunk entirely
+ if the loop count in the header did not match with the calculated count based on the size of the
+ chunk. When this happens, the second stage will still hit this path but the `pMetadata` will be
+ non-null, but will either be pointing at the very end of the allocation or at the start of another
+ chunk. We need to check the loop counts for consistency *before* dereferencing the pMetadata object
+ so it's consistent with how we do it in the first stage.
+ */
+ loopCount = drwav_bytes_to_u32(smplHeaderData + 28);
+ calculatedLoopCount = (pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES) / DRWAV_SMPL_LOOP_BYTES;
+ if (loopCount != calculatedLoopCount) {
+ return totalBytesRead;
+ }
+
+ pMetadata->type = drwav_metadata_type_smpl;
+ pMetadata->data.smpl.manufacturerId = drwav_bytes_to_u32(smplHeaderData + 0);
+ pMetadata->data.smpl.productId = drwav_bytes_to_u32(smplHeaderData + 4);
+ pMetadata->data.smpl.samplePeriodNanoseconds = drwav_bytes_to_u32(smplHeaderData + 8);
+ pMetadata->data.smpl.midiUnityNote = drwav_bytes_to_u32(smplHeaderData + 12);
+ pMetadata->data.smpl.midiPitchFraction = drwav_bytes_to_u32(smplHeaderData + 16);
+ pMetadata->data.smpl.smpteFormat = drwav_bytes_to_u32(smplHeaderData + 20);
+ pMetadata->data.smpl.smpteOffset = drwav_bytes_to_u32(smplHeaderData + 24);
+ pMetadata->data.smpl.sampleLoopCount = drwav_bytes_to_u32(smplHeaderData + 28);
+ pMetadata->data.smpl.samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(smplHeaderData + 32);
+
+ /*
+ The loop count needs to be validated against the size of the chunk for safety so we don't
+ attempt to read over the boundary of the chunk.
+ */
+ if (pMetadata->data.smpl.sampleLoopCount == calculatedLoopCount) {
+ pMetadata->data.smpl.pLoops = (drwav_smpl_loop*)drwav__metadata_get_memory(pParser, sizeof(drwav_smpl_loop) * pMetadata->data.smpl.sampleLoopCount, DRWAV_METADATA_ALIGNMENT);
+
+ for (iSampleLoop = 0; iSampleLoop < pMetadata->data.smpl.sampleLoopCount; ++iSampleLoop) {
+ drwav_uint8 smplLoopData[DRWAV_SMPL_LOOP_BYTES];
+ bytesJustRead = drwav__metadata_parser_read(pParser, smplLoopData, sizeof(smplLoopData), &totalBytesRead);
+
+ if (bytesJustRead == sizeof(smplLoopData)) {
+ pMetadata->data.smpl.pLoops[iSampleLoop].cuePointId = drwav_bytes_to_u32(smplLoopData + 0);
+ pMetadata->data.smpl.pLoops[iSampleLoop].type = drwav_bytes_to_u32(smplLoopData + 4);
+ pMetadata->data.smpl.pLoops[iSampleLoop].firstSampleOffset = drwav_bytes_to_u32(smplLoopData + 8);
+ pMetadata->data.smpl.pLoops[iSampleLoop].lastSampleOffset = drwav_bytes_to_u32(smplLoopData + 12);
+ pMetadata->data.smpl.pLoops[iSampleLoop].sampleFraction = drwav_bytes_to_u32(smplLoopData + 16);
+ pMetadata->data.smpl.pLoops[iSampleLoop].playCount = drwav_bytes_to_u32(smplLoopData + 20);
+ } else {
+ break;
+ }
+ }
+
+ if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) {
+ pMetadata->data.smpl.pSamplerSpecificData = drwav__metadata_get_memory(pParser, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, 1);
+ DRWAV_ASSERT(pMetadata->data.smpl.pSamplerSpecificData != NULL);
+
+ drwav__metadata_parser_read(pParser, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, &totalBytesRead);
+ }
+ } else {
+ /*
+ Getting here means the loop count in the header does not match up with the size of the
+ chunk. Clear out the data to zero just to be safe.
+
+ This should never actually get hit because we check for it above, but keeping this here
+ for added safety.
+ */
+ DRWAV_ZERO_OBJECT(&pMetadata->data.smpl);
+ }
+ }
+
+ return totalBytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_cue_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata)
+{
+ drwav_uint8 cueHeaderSectionData[DRWAV_CUE_BYTES];
+ drwav_uint64 totalBytesRead = 0;
+ size_t bytesJustRead;
+
+ if (pMetadata == NULL) {
+ return 0;
+ }
+
+ bytesJustRead = drwav__metadata_parser_read(pParser, cueHeaderSectionData, sizeof(cueHeaderSectionData), &totalBytesRead);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+
+ if (bytesJustRead == sizeof(cueHeaderSectionData)) {
+ pMetadata->type = drwav_metadata_type_cue;
+ pMetadata->data.cue.cuePointCount = drwav_bytes_to_u32(cueHeaderSectionData);
+
+ /*
+ We need to validate the cue point count against the size of the chunk so we don't read
+ beyond the chunk.
+ */
+ if (pMetadata->data.cue.cuePointCount == (pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES) {
+ pMetadata->data.cue.pCuePoints = (drwav_cue_point*)drwav__metadata_get_memory(pParser, sizeof(drwav_cue_point) * pMetadata->data.cue.cuePointCount, DRWAV_METADATA_ALIGNMENT);
+ DRWAV_ASSERT(pMetadata->data.cue.pCuePoints != NULL);
+
+ if (pMetadata->data.cue.cuePointCount > 0) {
+ drwav_uint32 iCuePoint;
+
+ for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) {
+ drwav_uint8 cuePointData[DRWAV_CUE_POINT_BYTES];
+ bytesJustRead = drwav__metadata_parser_read(pParser, cuePointData, sizeof(cuePointData), &totalBytesRead);
+
+ if (bytesJustRead == sizeof(cuePointData)) {
+ pMetadata->data.cue.pCuePoints[iCuePoint].id = drwav_bytes_to_u32(cuePointData + 0);
+ pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition = drwav_bytes_to_u32(cuePointData + 4);
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[0] = cuePointData[8];
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[1] = cuePointData[9];
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[2] = cuePointData[10];
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[3] = cuePointData[11];
+ pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart = drwav_bytes_to_u32(cuePointData + 12);
+ pMetadata->data.cue.pCuePoints[iCuePoint].blockStart = drwav_bytes_to_u32(cuePointData + 16);
+ pMetadata->data.cue.pCuePoints[iCuePoint].sampleOffset = drwav_bytes_to_u32(cuePointData + 20);
+ } else {
+ break;
+ }
+ }
+ }
+ }
+ }
+
+ return totalBytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_inst_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata)
+{
+ drwav_uint8 instData[DRWAV_INST_BYTES];
+ drwav_uint64 bytesRead;
+
+ if (pMetadata == NULL) {
+ return 0;
+ }
+
+ bytesRead = drwav__metadata_parser_read(pParser, instData, sizeof(instData), NULL);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+
+ if (bytesRead == sizeof(instData)) {
+ pMetadata->type = drwav_metadata_type_inst;
+ pMetadata->data.inst.midiUnityNote = (drwav_int8)instData[0];
+ pMetadata->data.inst.fineTuneCents = (drwav_int8)instData[1];
+ pMetadata->data.inst.gainDecibels = (drwav_int8)instData[2];
+ pMetadata->data.inst.lowNote = (drwav_int8)instData[3];
+ pMetadata->data.inst.highNote = (drwav_int8)instData[4];
+ pMetadata->data.inst.lowVelocity = (drwav_int8)instData[5];
+ pMetadata->data.inst.highVelocity = (drwav_int8)instData[6];
+ }
+
+ return bytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_acid_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata)
+{
+ drwav_uint8 acidData[DRWAV_ACID_BYTES];
+ drwav_uint64 bytesRead;
+
+ if (pMetadata == NULL) {
+ return 0;
+ }
+
+ bytesRead = drwav__metadata_parser_read(pParser, acidData, sizeof(acidData), NULL);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+
+ if (bytesRead == sizeof(acidData)) {
+ pMetadata->type = drwav_metadata_type_acid;
+ pMetadata->data.acid.flags = drwav_bytes_to_u32(acidData + 0);
+ pMetadata->data.acid.midiUnityNote = drwav_bytes_to_u16(acidData + 4);
+ pMetadata->data.acid.reserved1 = drwav_bytes_to_u16(acidData + 6);
+ pMetadata->data.acid.reserved2 = drwav_bytes_to_f32(acidData + 8);
+ pMetadata->data.acid.numBeats = drwav_bytes_to_u32(acidData + 12);
+ pMetadata->data.acid.meterDenominator = drwav_bytes_to_u16(acidData + 16);
+ pMetadata->data.acid.meterNumerator = drwav_bytes_to_u16(acidData + 18);
+ pMetadata->data.acid.tempo = drwav_bytes_to_f32(acidData + 20);
+ }
+
+ return bytesRead;
+}
+
+DRWAV_PRIVATE size_t drwav__strlen(const char* str)
+{
+ size_t result = 0;
+
+ while (*str++) {
+ result += 1;
+ }
+
+ return result;
+}
+
+DRWAV_PRIVATE size_t drwav__strlen_clamped(const char* str, size_t maxToRead)
+{
+ size_t result = 0;
+
+ while (*str++ && result < maxToRead) {
+ result += 1;
+ }
+
+ return result;
+}
+
+DRWAV_PRIVATE char* drwav__metadata_copy_string(drwav__metadata_parser* pParser, const char* str, size_t maxToRead)
+{
+ size_t len = drwav__strlen_clamped(str, maxToRead);
+
+ if (len) {
+ char* result = (char*)drwav__metadata_get_memory(pParser, len + 1, 1);
+ DRWAV_ASSERT(result != NULL);
+
+ DRWAV_COPY_MEMORY(result, str, len);
+ result[len] = '\0';
+
+ return result;
+ } else {
+ return NULL;
+ }
+}
+
+typedef struct
+{
+ const void* pBuffer;
+ size_t sizeInBytes;
+ size_t cursor;
+} drwav_buffer_reader;
+
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_init(const void* pBuffer, size_t sizeInBytes, drwav_buffer_reader* pReader)
+{
+ DRWAV_ASSERT(pBuffer != NULL);
+ DRWAV_ASSERT(pReader != NULL);
+
+ DRWAV_ZERO_OBJECT(pReader);
+
+ pReader->pBuffer = pBuffer;
+ pReader->sizeInBytes = sizeInBytes;
+ pReader->cursor = 0;
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_PRIVATE const void* drwav_buffer_reader_ptr(const drwav_buffer_reader* pReader)
+{
+ DRWAV_ASSERT(pReader != NULL);
+
+ return drwav_offset_ptr(pReader->pBuffer, pReader->cursor);
+}
+
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_seek(drwav_buffer_reader* pReader, size_t bytesToSeek)
+{
+ DRWAV_ASSERT(pReader != NULL);
+
+ if (pReader->cursor + bytesToSeek > pReader->sizeInBytes) {
+ return DRWAV_BAD_SEEK; /* Seeking too far forward. */
+ }
+
+ pReader->cursor += bytesToSeek;
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_read(drwav_buffer_reader* pReader, void* pDst, size_t bytesToRead, size_t* pBytesRead)
+{
+ drwav_result result = DRWAV_SUCCESS;
+ size_t bytesRemaining;
+
+ DRWAV_ASSERT(pReader != NULL);
+
+ if (pBytesRead != NULL) {
+ *pBytesRead = 0;
+ }
+
+ bytesRemaining = (pReader->sizeInBytes - pReader->cursor);
+ if (bytesToRead > bytesRemaining) {
+ bytesToRead = bytesRemaining;
+ }
+
+ if (pDst == NULL) {
+ /* Seek. */
+ result = drwav_buffer_reader_seek(pReader, bytesToRead);
+ } else {
+ /* Read. */
+ DRWAV_COPY_MEMORY(pDst, drwav_buffer_reader_ptr(pReader), bytesToRead);
+ pReader->cursor += bytesToRead;
+ }
+
+ DRWAV_ASSERT(pReader->cursor <= pReader->sizeInBytes);
+
+ if (result == DRWAV_SUCCESS) {
+ if (pBytesRead != NULL) {
+ *pBytesRead = bytesToRead;
+ }
+ }
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u16(drwav_buffer_reader* pReader, drwav_uint16* pDst)
+{
+ drwav_result result;
+ size_t bytesRead;
+ drwav_uint8 data[2];
+
+ DRWAV_ASSERT(pReader != NULL);
+ DRWAV_ASSERT(pDst != NULL);
+
+ *pDst = 0; /* Safety. */
+
+ result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead);
+ if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) {
+ return result;
+ }
+
+ *pDst = drwav_bytes_to_u16(data);
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u32(drwav_buffer_reader* pReader, drwav_uint32* pDst)
+{
+ drwav_result result;
+ size_t bytesRead;
+ drwav_uint8 data[4];
+
+ DRWAV_ASSERT(pReader != NULL);
+ DRWAV_ASSERT(pDst != NULL);
+
+ *pDst = 0; /* Safety. */
+
+ result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead);
+ if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) {
+ return result;
+ }
+
+ *pDst = drwav_bytes_to_u32(data);
+
+ return DRWAV_SUCCESS;
+}
+
+
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_bext_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize)
+{
+ drwav_uint8 bextData[DRWAV_BEXT_BYTES];
+ size_t bytesRead = drwav__metadata_parser_read(pParser, bextData, sizeof(bextData), NULL);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+
+ if (bytesRead == sizeof(bextData)) {
+ drwav_buffer_reader reader;
+ drwav_uint32 timeReferenceLow;
+ drwav_uint32 timeReferenceHigh;
+ size_t extraBytes;
+
+ pMetadata->type = drwav_metadata_type_bext;
+
+ if (drwav_buffer_reader_init(bextData, bytesRead, &reader) == DRWAV_SUCCESS) {
+ pMetadata->data.bext.pDescription = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_DESCRIPTION_BYTES);
+ drwav_buffer_reader_seek(&reader, DRWAV_BEXT_DESCRIPTION_BYTES);
+
+ pMetadata->data.bext.pOriginatorName = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_NAME_BYTES);
+ drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_NAME_BYTES);
+
+ pMetadata->data.bext.pOriginatorReference = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_REF_BYTES);
+ drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_REF_BYTES);
+
+ drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate), NULL);
+ drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime), NULL);
+
+ drwav_buffer_reader_read_u32(&reader, &timeReferenceLow);
+ drwav_buffer_reader_read_u32(&reader, &timeReferenceHigh);
+ pMetadata->data.bext.timeReference = ((drwav_uint64)timeReferenceHigh << 32) + timeReferenceLow;
+
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.version);
+
+ pMetadata->data.bext.pUMID = drwav__metadata_get_memory(pParser, DRWAV_BEXT_UMID_BYTES, 1);
+ drwav_buffer_reader_read(&reader, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES, NULL);
+
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessValue);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessRange);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxTruePeakLevel);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxMomentaryLoudness);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxShortTermLoudness);
+
+ DRWAV_ASSERT((drwav_offset_ptr(drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_RESERVED_BYTES)) == (bextData + DRWAV_BEXT_BYTES));
+
+ extraBytes = (size_t)(chunkSize - DRWAV_BEXT_BYTES);
+ if (extraBytes > 0) {
+ pMetadata->data.bext.pCodingHistory = (char*)drwav__metadata_get_memory(pParser, extraBytes + 1, 1);
+ DRWAV_ASSERT(pMetadata->data.bext.pCodingHistory != NULL);
+
+ bytesRead += drwav__metadata_parser_read(pParser, pMetadata->data.bext.pCodingHistory, extraBytes, NULL);
+ pMetadata->data.bext.codingHistorySize = (drwav_uint32)drwav__strlen(pMetadata->data.bext.pCodingHistory);
+ } else {
+ pMetadata->data.bext.pCodingHistory = NULL;
+ pMetadata->data.bext.codingHistorySize = 0;
+ }
+ }
+ }
+
+ return bytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_list_label_or_note_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize, drwav_metadata_type type)
+{
+ drwav_uint8 cueIDBuffer[DRWAV_LIST_LABEL_OR_NOTE_BYTES];
+ drwav_uint64 totalBytesRead = 0;
+ size_t bytesJustRead = drwav__metadata_parser_read(pParser, cueIDBuffer, sizeof(cueIDBuffer), &totalBytesRead);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+
+ if (bytesJustRead == sizeof(cueIDBuffer)) {
+ drwav_uint32 sizeIncludingNullTerminator;
+
+ pMetadata->type = type;
+ pMetadata->data.labelOrNote.cuePointId = drwav_bytes_to_u32(cueIDBuffer);
+
+ sizeIncludingNullTerminator = (drwav_uint32)chunkSize - DRWAV_LIST_LABEL_OR_NOTE_BYTES;
+ if (sizeIncludingNullTerminator > 0) {
+ pMetadata->data.labelOrNote.stringLength = sizeIncludingNullTerminator - 1;
+ pMetadata->data.labelOrNote.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1);
+ DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL);
+
+ drwav__metadata_parser_read(pParser, pMetadata->data.labelOrNote.pString, sizeIncludingNullTerminator, &totalBytesRead);
+ } else {
+ pMetadata->data.labelOrNote.stringLength = 0;
+ pMetadata->data.labelOrNote.pString = NULL;
+ }
+ }
+
+ return totalBytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__read_list_labelled_cue_region_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize)
+{
+ drwav_uint8 buffer[DRWAV_LIST_LABELLED_TEXT_BYTES];
+ drwav_uint64 totalBytesRead = 0;
+ size_t bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &totalBytesRead);
+
+ DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+
+ if (bytesJustRead == sizeof(buffer)) {
+ drwav_uint32 sizeIncludingNullTerminator;
+
+ pMetadata->type = drwav_metadata_type_list_labelled_cue_region;
+ pMetadata->data.labelledCueRegion.cuePointId = drwav_bytes_to_u32(buffer + 0);
+ pMetadata->data.labelledCueRegion.sampleLength = drwav_bytes_to_u32(buffer + 4);
+ pMetadata->data.labelledCueRegion.purposeId[0] = buffer[8];
+ pMetadata->data.labelledCueRegion.purposeId[1] = buffer[9];
+ pMetadata->data.labelledCueRegion.purposeId[2] = buffer[10];
+ pMetadata->data.labelledCueRegion.purposeId[3] = buffer[11];
+ pMetadata->data.labelledCueRegion.country = drwav_bytes_to_u16(buffer + 12);
+ pMetadata->data.labelledCueRegion.language = drwav_bytes_to_u16(buffer + 14);
+ pMetadata->data.labelledCueRegion.dialect = drwav_bytes_to_u16(buffer + 16);
+ pMetadata->data.labelledCueRegion.codePage = drwav_bytes_to_u16(buffer + 18);
+
+ sizeIncludingNullTerminator = (drwav_uint32)chunkSize - DRWAV_LIST_LABELLED_TEXT_BYTES;
+ if (sizeIncludingNullTerminator > 0) {
+ pMetadata->data.labelledCueRegion.stringLength = sizeIncludingNullTerminator - 1;
+ pMetadata->data.labelledCueRegion.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1);
+ DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL);
+
+ drwav__metadata_parser_read(pParser, pMetadata->data.labelledCueRegion.pString, sizeIncludingNullTerminator, &totalBytesRead);
+ } else {
+ pMetadata->data.labelledCueRegion.stringLength = 0;
+ pMetadata->data.labelledCueRegion.pString = NULL;
+ }
+ }
+
+ return totalBytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_info_text_chunk(drwav__metadata_parser* pParser, drwav_uint64 chunkSize, drwav_metadata_type type)
+{
+ drwav_uint64 bytesRead = 0;
+ drwav_uint32 stringSizeWithNullTerminator = (drwav_uint32)chunkSize;
+
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ pParser->metadataCount += 1;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, stringSizeWithNullTerminator, 1);
+ } else {
+ drwav_metadata* pMetadata = &pParser->pMetadata[pParser->metadataCursor];
+ pMetadata->type = type;
+ if (stringSizeWithNullTerminator > 0) {
+ pMetadata->data.infoText.stringLength = stringSizeWithNullTerminator - 1;
+ pMetadata->data.infoText.pString = (char*)drwav__metadata_get_memory(pParser, stringSizeWithNullTerminator, 1);
+ DRWAV_ASSERT(pMetadata->data.infoText.pString != NULL);
+
+ bytesRead = drwav__metadata_parser_read(pParser, pMetadata->data.infoText.pString, (size_t)stringSizeWithNullTerminator, NULL);
+ if (bytesRead == chunkSize) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ } else {
+ pMetadata->data.infoText.stringLength = 0;
+ pMetadata->data.infoText.pString = NULL;
+ pParser->metadataCursor += 1;
+ }
+ }
+
+ return bytesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_unknown_chunk(drwav__metadata_parser* pParser, const drwav_uint8* pChunkId, drwav_uint64 chunkSize, drwav_metadata_location location)
+{
+ drwav_uint64 bytesRead = 0;
+
+ if (location == drwav_metadata_location_invalid) {
+ return 0;
+ }
+
+ if (drwav_fourcc_equal(pChunkId, "data") || drwav_fourcc_equal(pChunkId, "fmt ") || drwav_fourcc_equal(pChunkId, "fact")) {
+ return 0;
+ }
+
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ pParser->metadataCount += 1;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)chunkSize, 1);
+ } else {
+ drwav_metadata* pMetadata = &pParser->pMetadata[pParser->metadataCursor];
+ pMetadata->type = drwav_metadata_type_unknown;
+ pMetadata->data.unknown.chunkLocation = location;
+ pMetadata->data.unknown.id[0] = pChunkId[0];
+ pMetadata->data.unknown.id[1] = pChunkId[1];
+ pMetadata->data.unknown.id[2] = pChunkId[2];
+ pMetadata->data.unknown.id[3] = pChunkId[3];
+ pMetadata->data.unknown.dataSizeInBytes = (drwav_uint32)chunkSize;
+ pMetadata->data.unknown.pData = (drwav_uint8 *)drwav__metadata_get_memory(pParser, (size_t)chunkSize, 1);
+ DRWAV_ASSERT(pMetadata->data.unknown.pData != NULL);
+
+ bytesRead = drwav__metadata_parser_read(pParser, pMetadata->data.unknown.pData, pMetadata->data.unknown.dataSizeInBytes, NULL);
+ if (bytesRead == pMetadata->data.unknown.dataSizeInBytes) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to read. */
+ }
+ }
+
+ return bytesRead;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__chunk_matches(drwav_metadata_type allowedMetadataTypes, const drwav_uint8* pChunkID, drwav_metadata_type type, const char* pID)
+{
+ return (allowedMetadataTypes & type) && drwav_fourcc_equal(pChunkID, pID);
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata_type allowedMetadataTypes)
+{
+ const drwav_uint8 *pChunkID = pChunkHeader->id.fourcc;
+ drwav_uint64 bytesRead = 0;
+
+ if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_smpl, "smpl")) {
+ if (pChunkHeader->sizeInBytes >= DRWAV_SMPL_BYTES) {
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ drwav_uint8 buffer[4];
+ size_t bytesJustRead;
+
+ if (!pParser->onSeek(pParser->pReadSeekUserData, 28, DRWAV_SEEK_CUR)) {
+ return bytesRead;
+ }
+ bytesRead += 28;
+
+ bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead);
+ if (bytesJustRead == sizeof(buffer)) {
+ drwav_uint32 loopCount = drwav_bytes_to_u32(buffer);
+ drwav_uint64 calculatedLoopCount;
+
+ /* The loop count must be validated against the size of the chunk. */
+ calculatedLoopCount = (pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES) / DRWAV_SMPL_LOOP_BYTES;
+ if (calculatedLoopCount == loopCount) {
+ bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead);
+ if (bytesJustRead == sizeof(buffer)) {
+ drwav_uint32 samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(buffer);
+
+ pParser->metadataCount += 1;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_smpl_loop) * loopCount, DRWAV_METADATA_ALIGNMENT);
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, samplerSpecificDataSizeInBytes, 1);
+ }
+ } else {
+ /* Loop count in header does not match the size of the chunk. */
+ }
+ }
+ } else {
+ bytesRead = drwav__read_smpl_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]);
+ if (bytesRead == pChunkHeader->sizeInBytes) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_inst, "inst")) {
+ if (pChunkHeader->sizeInBytes == DRWAV_INST_BYTES) {
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ pParser->metadataCount += 1;
+ } else {
+ bytesRead = drwav__read_inst_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]);
+ if (bytesRead == pChunkHeader->sizeInBytes) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_acid, "acid")) {
+ if (pChunkHeader->sizeInBytes == DRWAV_ACID_BYTES) {
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ pParser->metadataCount += 1;
+ } else {
+ bytesRead = drwav__read_acid_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]);
+ if (bytesRead == pChunkHeader->sizeInBytes) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_cue, "cue ")) {
+ if (pChunkHeader->sizeInBytes >= DRWAV_CUE_BYTES) {
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ size_t cueCount;
+
+ pParser->metadataCount += 1;
+ cueCount = (size_t)(pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_cue_point) * cueCount, DRWAV_METADATA_ALIGNMENT);
+ } else {
+ bytesRead = drwav__read_cue_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]);
+ if (bytesRead == pChunkHeader->sizeInBytes) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_bext, "bext")) {
+ if (pChunkHeader->sizeInBytes >= DRWAV_BEXT_BYTES) {
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ /* The description field is the largest one in a bext chunk, so that is the max size of this temporary buffer. */
+ char buffer[DRWAV_BEXT_DESCRIPTION_BYTES + 1];
+ size_t allocSizeNeeded = DRWAV_BEXT_UMID_BYTES; /* We know we will need SMPTE umid size. */
+ size_t bytesJustRead;
+
+ buffer[DRWAV_BEXT_DESCRIPTION_BYTES] = '\0';
+ bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_DESCRIPTION_BYTES, &bytesRead);
+ if (bytesJustRead != DRWAV_BEXT_DESCRIPTION_BYTES) {
+ return bytesRead;
+ }
+ allocSizeNeeded += drwav__strlen(buffer) + 1;
+
+ buffer[DRWAV_BEXT_ORIGINATOR_NAME_BYTES] = '\0';
+ bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_ORIGINATOR_NAME_BYTES, &bytesRead);
+ if (bytesJustRead != DRWAV_BEXT_ORIGINATOR_NAME_BYTES) {
+ return bytesRead;
+ }
+ allocSizeNeeded += drwav__strlen(buffer) + 1;
+
+ buffer[DRWAV_BEXT_ORIGINATOR_REF_BYTES] = '\0';
+ bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_ORIGINATOR_REF_BYTES, &bytesRead);
+ if (bytesJustRead != DRWAV_BEXT_ORIGINATOR_REF_BYTES) {
+ return bytesRead;
+ }
+ allocSizeNeeded += drwav__strlen(buffer) + 1;
+ allocSizeNeeded += (size_t)pChunkHeader->sizeInBytes - DRWAV_BEXT_BYTES + 1; /* Coding history. */
+
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, allocSizeNeeded, 1);
+
+ pParser->metadataCount += 1;
+ } else {
+ bytesRead = drwav__read_bext_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], pChunkHeader->sizeInBytes);
+ if (bytesRead == pChunkHeader->sizeInBytes) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav_fourcc_equal(pChunkID, "LIST") || drwav_fourcc_equal(pChunkID, "list")) {
+ drwav_metadata_location listType = drwav_metadata_location_invalid;
+ while (bytesRead < pChunkHeader->sizeInBytes) {
+ drwav_uint8 subchunkId[4];
+ drwav_uint8 subchunkSizeBuffer[4];
+ drwav_uint64 subchunkDataSize;
+ drwav_uint64 subchunkBytesRead = 0;
+ drwav_uint64 bytesJustRead = drwav__metadata_parser_read(pParser, subchunkId, sizeof(subchunkId), &bytesRead);
+ if (bytesJustRead != sizeof(subchunkId)) {
+ break;
+ }
+
+ /*
+ The first thing in a list chunk should be "adtl" or "INFO".
+
+ - adtl means this list is a Associated Data List Chunk and will contain labels, notes
+ or labelled cue regions.
+ - INFO means this list is an Info List Chunk containing info text chunks such as IPRD
+ which would specifies the album of this wav file.
+
+ No data follows the adtl or INFO id so we just make note of what type this list is and
+ continue.
+ */
+ if (drwav_fourcc_equal(subchunkId, "adtl")) {
+ listType = drwav_metadata_location_inside_adtl_list;
+ continue;
+ } else if (drwav_fourcc_equal(subchunkId, "INFO")) {
+ listType = drwav_metadata_location_inside_info_list;
+ continue;
+ }
+
+ bytesJustRead = drwav__metadata_parser_read(pParser, subchunkSizeBuffer, sizeof(subchunkSizeBuffer), &bytesRead);
+ if (bytesJustRead != sizeof(subchunkSizeBuffer)) {
+ break;
+ }
+ subchunkDataSize = drwav_bytes_to_u32(subchunkSizeBuffer);
+
+ if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_label, "labl") || drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_note, "note")) {
+ if (subchunkDataSize >= DRWAV_LIST_LABEL_OR_NOTE_BYTES) {
+ drwav_uint64 stringSizeWithNullTerm = subchunkDataSize - DRWAV_LIST_LABEL_OR_NOTE_BYTES;
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ pParser->metadataCount += 1;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerm, 1);
+ } else {
+ subchunkBytesRead = drwav__read_list_label_or_note_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize, drwav_fourcc_equal(subchunkId, "labl") ? drwav_metadata_type_list_label : drwav_metadata_type_list_note);
+ if (subchunkBytesRead == subchunkDataSize) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_labelled_cue_region, "ltxt")) {
+ if (subchunkDataSize >= DRWAV_LIST_LABELLED_TEXT_BYTES) {
+ drwav_uint64 stringSizeWithNullTerminator = subchunkDataSize - DRWAV_LIST_LABELLED_TEXT_BYTES;
+ if (pParser->stage == drwav__metadata_parser_stage_count) {
+ pParser->metadataCount += 1;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerminator, 1);
+ } else {
+ subchunkBytesRead = drwav__read_list_labelled_cue_region_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize);
+ if (subchunkBytesRead == subchunkDataSize) {
+ pParser->metadataCursor += 1;
+ } else {
+ /* Failed to parse. */
+ }
+ }
+ } else {
+ /* Incorrectly formed chunk. */
+ }
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_software, "ISFT")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_software);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_copyright, "ICOP")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_copyright);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_title, "INAM")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_title);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_artist, "IART")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_artist);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_comment, "ICMT")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_comment);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_date, "ICRD")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_date);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_genre, "IGNR")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_genre);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_album, "IPRD")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_album);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_tracknumber, "ITRK")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_tracknumber);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_location, "IARL")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_location);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_organization, "ICMS")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_organization);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_keywords, "IKEY")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_keywords);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_medium, "IMED")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_medium);
+ } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_description, "ISBJ")) {
+ subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_description);
+ } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) {
+ subchunkBytesRead = drwav__metadata_process_unknown_chunk(pParser, subchunkId, subchunkDataSize, listType);
+ }
+
+ bytesRead += subchunkBytesRead;
+ DRWAV_ASSERT(subchunkBytesRead <= subchunkDataSize);
+
+ if (subchunkBytesRead < subchunkDataSize) {
+ drwav_uint64 bytesToSeek = subchunkDataSize - subchunkBytesRead;
+
+ if (!pParser->onSeek(pParser->pReadSeekUserData, (int)bytesToSeek, DRWAV_SEEK_CUR)) {
+ break;
+ }
+ bytesRead += bytesToSeek;
+ }
+
+ if ((subchunkDataSize % 2) == 1) {
+ if (!pParser->onSeek(pParser->pReadSeekUserData, 1, DRWAV_SEEK_CUR)) {
+ break;
+ }
+ bytesRead += 1;
+ }
+ }
+ } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) {
+ bytesRead = drwav__metadata_process_unknown_chunk(pParser, pChunkID, pChunkHeader->sizeInBytes, drwav_metadata_location_top_level);
+ }
+
+ return bytesRead;
+}
+
+
+DRWAV_PRIVATE drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav)
+{
+ drwav_uint32 bytesPerFrame;
+
+ /*
+ The bytes per frame is a bit ambiguous. It can be either be based on the bits per sample, or the block align. The way I'm doing it here
+ is that if the bits per sample is a multiple of 8, use floor(bitsPerSample*channels/8), otherwise fall back to the block align.
+ */
+ if ((pWav->bitsPerSample & 0x7) == 0) {
+ /* Bits per sample is a multiple of 8. */
+ bytesPerFrame = (pWav->bitsPerSample * pWav->fmt.channels) >> 3;
+ } else {
+ bytesPerFrame = pWav->fmt.blockAlign;
+ }
+
+ /* Validation for known formats. a-law and mu-law should be 1 byte per channel. If it's not, it's not decodable. */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW || pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
+ if (bytesPerFrame != pWav->fmt.channels) {
+ return 0; /* Invalid file. */
+ }
+ }
+
+ return bytesPerFrame;
+}
+
+DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT)
+{
+ if (pFMT == NULL) {
+ return 0;
+ }
+
+ if (pFMT->formatTag != DR_WAVE_FORMAT_EXTENSIBLE) {
+ return pFMT->formatTag;
+ } else {
+ return drwav_bytes_to_u16(pFMT->subFormat); /* Only the first two bytes are required. */
+ }
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav_preinit(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pReadSeekTellUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pWav == NULL || onRead == NULL || onSeek == NULL) { /* <-- onTell is optional. */
+ return DRWAV_FALSE;
+ }
+
+ DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav));
+ pWav->onRead = onRead;
+ pWav->onSeek = onSeek;
+ pWav->onTell = onTell;
+ pWav->pUserData = pReadSeekTellUserData;
+ pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks);
+
+ if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) {
+ return DRWAV_FALSE; /* Invalid allocation callbacks. */
+ }
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags)
+{
+ /* This function assumes drwav_preinit() has been called beforehand. */
+ drwav_result result;
+ drwav_uint64 cursor; /* <-- Keeps track of the byte position so we can seek to specific locations. */
+ drwav_bool32 sequential;
+ drwav_uint8 riff[4];
+ drwav_fmt fmt;
+ unsigned short translatedFormatTag;
+ drwav_uint64 dataChunkSize = 0; /* <-- Important! Don't explicitly set this to 0 anywhere else. Calculation of the size of the data chunk is performed in different paths depending on the container. */
+ drwav_uint64 sampleCountFromFactChunk = 0; /* Same as dataChunkSize - make sure this is the only place this is initialized to 0. */
+ drwav_uint64 metadataStartPos;
+ drwav__metadata_parser metadataParser;
+ drwav_bool8 isProcessingMetadata = DRWAV_FALSE;
+ drwav_bool8 foundChunk_fmt = DRWAV_FALSE;
+ drwav_bool8 foundChunk_data = DRWAV_FALSE;
+ drwav_bool8 isAIFCFormType = DRWAV_FALSE; /* Only used with AIFF. */
+ drwav_uint64 aiffFrameCount = 0;
+
+ cursor = 0;
+ sequential = (flags & DRWAV_SEQUENTIAL) != 0;
+ DRWAV_ZERO_OBJECT(&fmt);
+
+ /* The first 4 bytes should be the RIFF identifier. */
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, riff, sizeof(riff), &cursor) != sizeof(riff)) {
+ return DRWAV_FALSE;
+ }
+
+ /*
+ The first 4 bytes can be used to identify the container. For RIFF files it will start with "RIFF" and for
+ w64 it will start with "riff".
+ */
+ if (drwav_fourcc_equal(riff, "RIFF")) {
+ pWav->container = drwav_container_riff;
+ } else if (drwav_fourcc_equal(riff, "RIFX")) {
+ pWav->container = drwav_container_rifx;
+ } else if (drwav_fourcc_equal(riff, "riff")) {
+ int i;
+ drwav_uint8 riff2[12];
+
+ pWav->container = drwav_container_w64;
+
+ /* Check the rest of the GUID for validity. */
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, riff2, sizeof(riff2), &cursor) != sizeof(riff2)) {
+ return DRWAV_FALSE;
+ }
+
+ for (i = 0; i < 12; ++i) {
+ if (riff2[i] != drwavGUID_W64_RIFF[i+4]) {
+ return DRWAV_FALSE;
+ }
+ }
+ } else if (drwav_fourcc_equal(riff, "RF64")) {
+ pWav->container = drwav_container_rf64;
+ } else if (drwav_fourcc_equal(riff, "FORM")) {
+ pWav->container = drwav_container_aiff;
+ } else {
+ return DRWAV_FALSE; /* Unknown or unsupported container. */
+ }
+
+
+ if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) {
+ drwav_uint8 chunkSizeBytes[4];
+ drwav_uint8 wave[4];
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) {
+ return DRWAV_FALSE;
+ }
+
+ if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx) {
+ if (drwav_bytes_to_u32_ex(chunkSizeBytes, pWav->container) < 36) {
+ /*
+ I've had a report of a WAV file failing to load when the size of the WAVE chunk is not encoded
+ and is instead just set to 0. I'm going to relax the validation here to allow these files to
+ load. Considering the chunk size isn't actually used this should be safe. With this change my
+ test suite still passes.
+ */
+ /*return DRWAV_FALSE;*/ /* Chunk size should always be at least 36 bytes. */
+ }
+ } else if (pWav->container == drwav_container_rf64) {
+ if (drwav_bytes_to_u32_le(chunkSizeBytes) != 0xFFFFFFFF) {
+ return DRWAV_FALSE; /* Chunk size should always be set to -1/0xFFFFFFFF for RF64. The actual size is retrieved later. */
+ }
+ } else {
+ return DRWAV_FALSE; /* Should never hit this. */
+ }
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) {
+ return DRWAV_FALSE;
+ }
+
+ if (!drwav_fourcc_equal(wave, "WAVE")) {
+ return DRWAV_FALSE; /* Expecting "WAVE". */
+ }
+ } else if (pWav->container == drwav_container_w64) {
+ drwav_uint8 chunkSizeBytes[8];
+ drwav_uint8 wave[16];
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) {
+ return DRWAV_FALSE;
+ }
+
+ if (drwav_bytes_to_u64(chunkSizeBytes) < 80) {
+ return DRWAV_FALSE;
+ }
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) {
+ return DRWAV_FALSE;
+ }
+
+ if (!drwav_guid_equal(wave, drwavGUID_W64_WAVE)) {
+ return DRWAV_FALSE;
+ }
+ } else if (pWav->container == drwav_container_aiff) {
+ drwav_uint8 chunkSizeBytes[4];
+ drwav_uint8 aiff[4];
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) {
+ return DRWAV_FALSE;
+ }
+
+ if (drwav_bytes_to_u32_be(chunkSizeBytes) < 18) {
+ return DRWAV_FALSE;
+ }
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, aiff, sizeof(aiff), &cursor) != sizeof(aiff)) {
+ return DRWAV_FALSE;
+ }
+
+ if (drwav_fourcc_equal(aiff, "AIFF")) {
+ isAIFCFormType = DRWAV_FALSE;
+ } else if (drwav_fourcc_equal(aiff, "AIFC")) {
+ isAIFCFormType = DRWAV_TRUE;
+ } else {
+ return DRWAV_FALSE; /* Expecting "AIFF" or "AIFC". */
+ }
+ } else {
+ return DRWAV_FALSE;
+ }
+
+
+ /* For RF64, the "ds64" chunk must come next, before the "fmt " chunk. */
+ if (pWav->container == drwav_container_rf64) {
+ drwav_uint8 sizeBytes[8];
+ drwav_uint64 bytesRemainingInChunk;
+ drwav_chunk_header header;
+ result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header);
+ if (result != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ if (!drwav_fourcc_equal(header.id.fourcc, "ds64")) {
+ return DRWAV_FALSE; /* Expecting "ds64". */
+ }
+
+ bytesRemainingInChunk = header.sizeInBytes + header.paddingSize;
+
+ /* We don't care about the size of the RIFF chunk - skip it. */
+ if (!drwav__seek_forward(pWav->onSeek, 8, pWav->pUserData)) {
+ return DRWAV_FALSE;
+ }
+ bytesRemainingInChunk -= 8;
+ cursor += 8;
+
+
+ /* Next 8 bytes is the size of the "data" chunk. */
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) {
+ return DRWAV_FALSE;
+ }
+ bytesRemainingInChunk -= 8;
+ dataChunkSize = drwav_bytes_to_u64(sizeBytes);
+
+
+ /* Next 8 bytes is the same count which we would usually derived from the FACT chunk if it was available. */
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) {
+ return DRWAV_FALSE;
+ }
+ bytesRemainingInChunk -= 8;
+ sampleCountFromFactChunk = drwav_bytes_to_u64(sizeBytes);
+
+
+ /* Skip over everything else. */
+ if (!drwav__seek_forward(pWav->onSeek, bytesRemainingInChunk, pWav->pUserData)) {
+ return DRWAV_FALSE;
+ }
+ cursor += bytesRemainingInChunk;
+ }
+
+
+ metadataStartPos = cursor;
+
+ /*
+ Whether or not we are processing metadata controls how we load. We can load more efficiently when
+ metadata is not being processed, but we also cannot process metadata for Wave64 because I have not
+ been able to test it. If someone is able to test this and provide a patch I'm happy to enable it.
+
+ Seqential mode cannot support metadata because it involves seeking backwards.
+ */
+ isProcessingMetadata = !sequential && ((flags & DRWAV_WITH_METADATA) != 0);
+
+ /* Don't allow processing of metadata with untested containers. */
+ if (pWav->container != drwav_container_riff && pWav->container != drwav_container_rf64) {
+ isProcessingMetadata = DRWAV_FALSE;
+ }
+
+ DRWAV_ZERO_MEMORY(&metadataParser, sizeof(metadataParser));
+ if (isProcessingMetadata) {
+ metadataParser.onRead = pWav->onRead;
+ metadataParser.onSeek = pWav->onSeek;
+ metadataParser.pReadSeekUserData = pWav->pUserData;
+ metadataParser.stage = drwav__metadata_parser_stage_count;
+ }
+
+
+ /*
+ From here on out, chunks might be in any order. In order to robustly handle metadata we'll need
+ to loop through every chunk and handle them as we find them. In sequential mode we need to get
+ out of the loop as soon as we find the data chunk because we won't be able to seek back.
+ */
+ for (;;) { /* For each chunk... */
+ drwav_chunk_header header;
+ drwav_uint64 chunkSize;
+
+ result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header);
+ if (result != DRWAV_SUCCESS) {
+ break;
+ }
+
+ chunkSize = header.sizeInBytes;
+
+
+ /*
+ Always tell the caller about this chunk. We cannot do this in sequential mode because the
+ callback is allowed to read from the file, in which case we'll need to rewind.
+ */
+ if (!sequential && onChunk != NULL) {
+ drwav_uint64 callbackBytesRead = onChunk(pChunkUserData, pWav->onRead, pWav->onSeek, pWav->pUserData, &header, pWav->container, &fmt);
+
+ /*
+ dr_wav may need to read the contents of the chunk, so we now need to seek back to the position before
+ we called the callback.
+ */
+ if (callbackBytesRead > 0) {
+ if (drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData) == DRWAV_FALSE) {
+ return DRWAV_FALSE;
+ }
+ }
+ }
+
+
+ /* Explicitly handle known chunks first. */
+
+ /* "fmt " */
+ if (((pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) && drwav_fourcc_equal(header.id.fourcc, "fmt ")) ||
+ ((pWav->container == drwav_container_w64) && drwav_guid_equal(header.id.guid, drwavGUID_W64_FMT))) {
+ drwav_uint8 fmtData[16];
+
+ foundChunk_fmt = DRWAV_TRUE;
+
+ if (pWav->onRead(pWav->pUserData, fmtData, sizeof(fmtData)) != sizeof(fmtData)) {
+ return DRWAV_FALSE;
+ }
+ cursor += sizeof(fmtData);
+
+ fmt.formatTag = drwav_bytes_to_u16_ex(fmtData + 0, pWav->container);
+ fmt.channels = drwav_bytes_to_u16_ex(fmtData + 2, pWav->container);
+ fmt.sampleRate = drwav_bytes_to_u32_ex(fmtData + 4, pWav->container);
+ fmt.avgBytesPerSec = drwav_bytes_to_u32_ex(fmtData + 8, pWav->container);
+ fmt.blockAlign = drwav_bytes_to_u16_ex(fmtData + 12, pWav->container);
+ fmt.bitsPerSample = drwav_bytes_to_u16_ex(fmtData + 14, pWav->container);
+
+ fmt.extendedSize = 0;
+ fmt.validBitsPerSample = 0;
+ fmt.channelMask = 0;
+ DRWAV_ZERO_MEMORY(fmt.subFormat, sizeof(fmt.subFormat));
+
+ if (header.sizeInBytes > 16) {
+ drwav_uint8 fmt_cbSize[2];
+ int bytesReadSoFar = 0;
+
+ if (pWav->onRead(pWav->pUserData, fmt_cbSize, sizeof(fmt_cbSize)) != sizeof(fmt_cbSize)) {
+ return DRWAV_FALSE; /* Expecting more data. */
+ }
+ cursor += sizeof(fmt_cbSize);
+
+ bytesReadSoFar = 18;
+
+ fmt.extendedSize = drwav_bytes_to_u16_ex(fmt_cbSize, pWav->container);
+ if (fmt.extendedSize > 0) {
+ /* Simple validation. */
+ if (fmt.formatTag == DR_WAVE_FORMAT_EXTENSIBLE) {
+ if (fmt.extendedSize != 22) {
+ return DRWAV_FALSE;
+ }
+ }
+
+ if (fmt.formatTag == DR_WAVE_FORMAT_EXTENSIBLE) {
+ drwav_uint8 fmtext[22];
+
+ if (pWav->onRead(pWav->pUserData, fmtext, fmt.extendedSize) != fmt.extendedSize) {
+ return DRWAV_FALSE; /* Expecting more data. */
+ }
+
+ fmt.validBitsPerSample = drwav_bytes_to_u16_ex(fmtext + 0, pWav->container);
+ fmt.channelMask = drwav_bytes_to_u32_ex(fmtext + 2, pWav->container);
+ drwav_bytes_to_guid(fmtext + 6, fmt.subFormat);
+ } else {
+ if (pWav->onSeek(pWav->pUserData, fmt.extendedSize, DRWAV_SEEK_CUR) == DRWAV_FALSE) {
+ return DRWAV_FALSE;
+ }
+ }
+ cursor += fmt.extendedSize;
+
+ bytesReadSoFar += fmt.extendedSize;
+ }
+
+ /* Seek past any leftover bytes. For w64 the leftover will be defined based on the chunk size. */
+ if (pWav->onSeek(pWav->pUserData, (int)(header.sizeInBytes - bytesReadSoFar), DRWAV_SEEK_CUR) == DRWAV_FALSE) {
+ return DRWAV_FALSE;
+ }
+ cursor += (header.sizeInBytes - bytesReadSoFar);
+ }
+
+ if (header.paddingSize > 0) {
+ if (drwav__seek_forward(pWav->onSeek, header.paddingSize, pWav->pUserData) == DRWAV_FALSE) {
+ break;
+ }
+ cursor += header.paddingSize;
+ }
+
+ /* Go to the next chunk. Don't include this chunk in metadata. */
+ continue;
+ }
+
+ /* "data" */
+ if (((pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) && drwav_fourcc_equal(header.id.fourcc, "data")) ||
+ ((pWav->container == drwav_container_w64) && drwav_guid_equal(header.id.guid, drwavGUID_W64_DATA))) {
+ foundChunk_data = DRWAV_TRUE;
+
+ pWav->dataChunkDataPos = cursor;
+
+ if (pWav->container != drwav_container_rf64) { /* The data chunk size for RF64 will always be set to 0xFFFFFFFF here. It was set to it's true value earlier. */
+ dataChunkSize = chunkSize;
+ }
+
+ /* If we're running in sequential mode, or we're not reading metadata, we have enough now that we can get out of the loop. */
+ if (sequential || !isProcessingMetadata) {
+ break; /* No need to keep reading beyond the data chunk. */
+ } else {
+ chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */
+ if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) {
+ break;
+ }
+ cursor += chunkSize;
+
+ continue; /* There may be some more metadata to read. */
+ }
+ }
+
+ /* "fact". This is optional. Can use this to get the sample count which is useful for compressed formats. For RF64 we retrieved the sample count from the ds64 chunk earlier. */
+ if (((pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) && drwav_fourcc_equal(header.id.fourcc, "fact")) ||
+ ((pWav->container == drwav_container_w64) && drwav_guid_equal(header.id.guid, drwavGUID_W64_FACT))) {
+ if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx) {
+ drwav_uint8 sampleCount[4];
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCount, 4, &cursor) != 4) {
+ return DRWAV_FALSE;
+ }
+
+ chunkSize -= 4;
+
+ /*
+ The sample count in the "fact" chunk is either unreliable, or I'm not understanding it properly. For now I am only enabling this
+ for Microsoft ADPCM formats.
+ */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
+ sampleCountFromFactChunk = drwav_bytes_to_u32_ex(sampleCount, pWav->container);
+ } else {
+ sampleCountFromFactChunk = 0;
+ }
+ } else if (pWav->container == drwav_container_w64) {
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCountFromFactChunk, 8, &cursor) != 8) {
+ return DRWAV_FALSE;
+ }
+
+ chunkSize -= 8;
+ } else if (pWav->container == drwav_container_rf64) {
+ /* We retrieved the sample count from the ds64 chunk earlier so no need to do that here. */
+ }
+
+ /* Seek to the next chunk in preparation for the next iteration. */
+ chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */
+ if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) {
+ break;
+ }
+ cursor += chunkSize;
+
+ continue;
+ }
+
+
+ /* "COMM". AIFF/AIFC only. */
+ if (pWav->container == drwav_container_aiff && drwav_fourcc_equal(header.id.fourcc, "COMM")) {
+ drwav_uint8 commData[24];
+ drwav_uint32 commDataBytesToRead;
+ drwav_uint16 channels;
+ drwav_uint32 frameCount;
+ drwav_uint16 sampleSizeInBits;
+ drwav_int64 sampleRate;
+ drwav_uint16 compressionFormat;
+
+ foundChunk_fmt = DRWAV_TRUE;
+
+ if (isAIFCFormType) {
+ commDataBytesToRead = 24;
+ if (header.sizeInBytes < commDataBytesToRead) {
+ return DRWAV_FALSE; /* Invalid COMM chunk. */
+ }
+ } else {
+ commDataBytesToRead = 18;
+ if (header.sizeInBytes != commDataBytesToRead) {
+ return DRWAV_FALSE; /* INVALID COMM chunk. */
+ }
+ }
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, commData, commDataBytesToRead, &cursor) != commDataBytesToRead) {
+ return DRWAV_FALSE;
+ }
+
+
+ channels = drwav_bytes_to_u16_ex (commData + 0, pWav->container);
+ frameCount = drwav_bytes_to_u32_ex (commData + 2, pWav->container);
+ sampleSizeInBits = drwav_bytes_to_u16_ex (commData + 6, pWav->container);
+ sampleRate = drwav_aiff_extented_to_s64(commData + 8);
+
+ if (sampleRate < 0 || sampleRate > 0xFFFFFFFF) {
+ return DRWAV_FALSE; /* Invalid sample rate. */
+ }
+
+ if (isAIFCFormType) {
+ const drwav_uint8* type = commData + 18;
+
+ if (drwav_fourcc_equal(type, "NONE")) {
+ compressionFormat = DR_WAVE_FORMAT_PCM; /* PCM, big-endian. */
+ } else if (drwav_fourcc_equal(type, "raw ")) {
+ compressionFormat = DR_WAVE_FORMAT_PCM;
+
+ /* In my testing, it looks like when the "raw " compression type is used, 8-bit samples should be considered unsigned. */
+ if (sampleSizeInBits == 8) {
+ pWav->aiff.isUnsigned = DRWAV_TRUE;
+ }
+ } else if (drwav_fourcc_equal(type, "sowt")) {
+ compressionFormat = DR_WAVE_FORMAT_PCM; /* PCM, little-endian. */
+ pWav->aiff.isLE = DRWAV_TRUE;
+ } else if (drwav_fourcc_equal(type, "fl32") || drwav_fourcc_equal(type, "fl64") || drwav_fourcc_equal(type, "FL32") || drwav_fourcc_equal(type, "FL64")) {
+ compressionFormat = DR_WAVE_FORMAT_IEEE_FLOAT;
+ } else if (drwav_fourcc_equal(type, "alaw") || drwav_fourcc_equal(type, "ALAW")) {
+ compressionFormat = DR_WAVE_FORMAT_ALAW;
+ } else if (drwav_fourcc_equal(type, "ulaw") || drwav_fourcc_equal(type, "ULAW")) {
+ compressionFormat = DR_WAVE_FORMAT_MULAW;
+ } else if (drwav_fourcc_equal(type, "ima4")) {
+ compressionFormat = DR_WAVE_FORMAT_DVI_ADPCM;
+ sampleSizeInBits = 4;
+
+ /*
+ I haven't been able to figure out how to get correct decoding for IMA ADPCM. Until this is figured out
+ we'll need to abort when we encounter such an encoding. Advice welcome!
+ */
+ (void)compressionFormat;
+ (void)sampleSizeInBits;
+
+ return DRWAV_FALSE;
+ } else {
+ return DRWAV_FALSE; /* Unknown or unsupported compression format. Need to abort. */
+ }
+ } else {
+ compressionFormat = DR_WAVE_FORMAT_PCM; /* It's a standard AIFF form which is always compressed. */
+ }
+
+ /* With AIFF we want to use the explicitly defined frame count rather than deriving it from the size of the chunk. */
+ aiffFrameCount = frameCount;
+
+ /* We should now have enough information to fill out our fmt structure. */
+ fmt.formatTag = compressionFormat;
+ fmt.channels = channels;
+ fmt.sampleRate = (drwav_uint32)sampleRate;
+ fmt.bitsPerSample = sampleSizeInBits;
+ fmt.blockAlign = (drwav_uint16)(fmt.channels * fmt.bitsPerSample / 8);
+ fmt.avgBytesPerSec = fmt.blockAlign * fmt.sampleRate;
+
+ if (fmt.blockAlign == 0 && compressionFormat == DR_WAVE_FORMAT_DVI_ADPCM) {
+ fmt.blockAlign = 34 * fmt.channels;
+ }
+
+ /*
+ Weird one. I've seen some alaw and ulaw encoded files that for some reason set the bits per sample to 16 when
+ it should be 8. To get this working I need to explicitly check for this and change it.
+ */
+ if (compressionFormat == DR_WAVE_FORMAT_ALAW || compressionFormat == DR_WAVE_FORMAT_MULAW) {
+ if (fmt.bitsPerSample > 8) {
+ fmt.bitsPerSample = 8;
+ fmt.blockAlign = fmt.channels;
+ }
+ }
+
+ /* In AIFF, samples are padded to 8 byte boundaries. We need to round up our bits per sample here. */
+ fmt.bitsPerSample += (fmt.bitsPerSample & 7);
+
+
+ /* If the form type is AIFC there will be some additional data in the chunk. We need to seek past it. */
+ if (isAIFCFormType) {
+ if (drwav__seek_forward(pWav->onSeek, (chunkSize - commDataBytesToRead), pWav->pUserData) == DRWAV_FALSE) {
+ return DRWAV_FALSE;
+ }
+ cursor += (chunkSize - commDataBytesToRead);
+ }
+
+ /* Don't fall through or else we'll end up treating this chunk as metadata which is incorrect. */
+ continue;
+ }
+
+
+ /* "SSND". AIFF/AIFC only. This is the AIFF equivalent of the "data" chunk. */
+ if (pWav->container == drwav_container_aiff && drwav_fourcc_equal(header.id.fourcc, "SSND")) {
+ drwav_uint8 offsetAndBlockSizeData[8];
+ drwav_uint32 offset;
+
+ foundChunk_data = DRWAV_TRUE;
+
+ if (drwav__on_read(pWav->onRead, pWav->pUserData, offsetAndBlockSizeData, sizeof(offsetAndBlockSizeData), &cursor) != sizeof(offsetAndBlockSizeData)) {
+ return DRWAV_FALSE;
+ }
+
+ /* The position of the audio data starts at an offset. */
+ offset = drwav_bytes_to_u32_ex(offsetAndBlockSizeData + 0, pWav->container);
+ pWav->dataChunkDataPos = cursor + offset;
+
+ /* The data chunk size needs to be reduced by the offset or else seeking will break. */
+ dataChunkSize = chunkSize;
+ if (dataChunkSize > offset) {
+ dataChunkSize -= offset;
+ } else {
+ dataChunkSize = 0;
+ }
+
+ if (sequential) {
+ if (foundChunk_fmt) { /* <-- Name is misleading, but will be set to true if the COMM chunk has been parsed. */
+ /*
+ Getting here means we're opening in sequential mode and we've found the SSND (data) and COMM (fmt) chunks. We need
+ to get out of the loop here or else we'll end up going past the data chunk and will have no way of getting back to
+ it since we're not allowed to seek backwards.
+
+ One subtle detail here is that there is an offset with the SSND chunk. We need to make sure we seek past this offset
+ so we're left sitting on the first byte of actual audio data.
+ */
+ if (drwav__seek_forward(pWav->onSeek, offset, pWav->pUserData) == DRWAV_FALSE) {
+ return DRWAV_FALSE;
+ }
+ cursor += offset;
+
+ break;
+ } else {
+ /*
+ Getting here means the COMM chunk was not found. In sequential mode, if we haven't yet found the COMM chunk
+ we'll need to abort because we can't be doing a backwards seek back to the SSND chunk in order to read the
+ data. For this reason, this configuration of AIFF files are not supported with sequential mode.
+ */
+ return DRWAV_FALSE;
+ }
+ } else {
+ chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */
+ chunkSize -= sizeof(offsetAndBlockSizeData); /* <-- This was read earlier. */
+
+ if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) {
+ break;
+ }
+ cursor += chunkSize;
+
+ continue; /* There may be some more metadata to read. */
+ }
+ }
+
+
+ /* Getting here means it's not a chunk that we care about internally, but might need to be handled as metadata by the caller. */
+ if (isProcessingMetadata) {
+ drwav__metadata_process_chunk(&metadataParser, &header, drwav_metadata_type_all_including_unknown);
+
+ /* Go back to the start of the chunk so we can normalize the position of the cursor. */
+ if (drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData) == DRWAV_FALSE) {
+ break; /* Failed to seek. Can't reliable read the remaining chunks. Get out. */
+ }
+ }
+
+
+ /* Make sure we skip past the content of this chunk before we go to the next one. */
+ chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */
+ if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) {
+ break;
+ }
+ cursor += chunkSize;
+ }
+
+ /* There's some mandatory chunks that must exist. If they were not found in the iteration above we must abort. */
+ if (!foundChunk_fmt || !foundChunk_data) {
+ return DRWAV_FALSE;
+ }
+
+ /* Basic validation. */
+ if ((fmt.sampleRate == 0 || fmt.sampleRate > DRWAV_MAX_SAMPLE_RATE ) ||
+ (fmt.channels == 0 || fmt.channels > DRWAV_MAX_CHANNELS ) ||
+ (fmt.bitsPerSample == 0 || fmt.bitsPerSample > DRWAV_MAX_BITS_PER_SAMPLE) ||
+ fmt.blockAlign == 0) {
+ return DRWAV_FALSE; /* Probably an invalid WAV file. */
+ }
+
+ /* Translate the internal format. */
+ translatedFormatTag = fmt.formatTag;
+ if (translatedFormatTag == DR_WAVE_FORMAT_EXTENSIBLE) {
+ translatedFormatTag = drwav_bytes_to_u16_ex(fmt.subFormat + 0, pWav->container);
+ }
+
+ /* We may have moved passed the data chunk. If so we need to move back. If running in sequential mode we can assume we are already sitting on the data chunk. */
+ if (!sequential) {
+ if (!drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData)) {
+ return DRWAV_FALSE;
+ }
+ cursor = pWav->dataChunkDataPos;
+ }
+
+
+ /*
+ At this point we should have done the initial parsing of each of our chunks, but we now need to
+ do a second pass to extract the actual contents of the metadata (the first pass just calculated
+ the length of the memory allocation).
+
+ We only do this if we've actually got metadata to parse.
+ */
+ if (isProcessingMetadata && metadataParser.metadataCount > 0) {
+ if (drwav__seek_from_start(pWav->onSeek, metadataStartPos, pWav->pUserData) == DRWAV_FALSE) {
+ return DRWAV_FALSE;
+ }
+
+ result = drwav__metadata_alloc(&metadataParser, &pWav->allocationCallbacks);
+ if (result != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ metadataParser.stage = drwav__metadata_parser_stage_read;
+
+ for (;;) {
+ drwav_chunk_header header;
+ drwav_uint64 metadataBytesRead;
+
+ result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header);
+ if (result != DRWAV_SUCCESS) {
+ break;
+ }
+
+ metadataBytesRead = drwav__metadata_process_chunk(&metadataParser, &header, drwav_metadata_type_all_including_unknown);
+
+ /* Move to the end of the chunk so we can keep iterating. */
+ if (drwav__seek_forward(pWav->onSeek, (header.sizeInBytes + header.paddingSize) - metadataBytesRead, pWav->pUserData) == DRWAV_FALSE) {
+ drwav_free(metadataParser.pMetadata, &pWav->allocationCallbacks);
+ return DRWAV_FALSE;
+ }
+ }
+
+ /* Getting here means we're finished parsing the metadata. */
+ pWav->pMetadata = metadataParser.pMetadata;
+ pWav->metadataCount = metadataParser.metadataCount;
+ }
+
+ /*
+ It's possible for the size reported in the data chunk to be greater than that of the file. We
+ need to do a validation check here to make sure we don't exceed the file size. To skip this
+ check, set the onTell callback to NULL.
+ */
+ if (pWav->onTell != NULL && pWav->onSeek != NULL) {
+ if (pWav->onSeek(pWav->pUserData, 0, DRWAV_SEEK_END) == DRWAV_TRUE) {
+ drwav_int64 fileSize;
+ if (pWav->onTell(pWav->pUserData, &fileSize)) {
+ if (dataChunkSize + pWav->dataChunkDataPos > (drwav_uint64)fileSize) {
+ dataChunkSize = (drwav_uint64)fileSize - pWav->dataChunkDataPos;
+ }
+ }
+ } else {
+ /*
+ Failed to seek to the end of the file. It might not be supported by the backend so in
+ this case we cannot perform the validation check.
+ */
+ }
+ }
+
+ /*
+ I've seen a WAV file in the wild where a RIFF-ecapsulated file has the size of it's "RIFF" and
+ "data" chunks set to 0xFFFFFFFF when the file is definitely not that big. In this case we're
+ going to have to calculate the size by reading and discarding bytes, and then seeking back. We
+ cannot do this in sequential mode. We just assume that the rest of the file is audio data.
+ */
+ if (dataChunkSize == 0xFFFFFFFF && (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx) && pWav->isSequentialWrite == DRWAV_FALSE) {
+ dataChunkSize = 0;
+
+ for (;;) {
+ drwav_uint8 temp[4096];
+ size_t bytesRead = pWav->onRead(pWav->pUserData, temp, sizeof(temp));
+ dataChunkSize += bytesRead;
+
+ if (bytesRead < sizeof(temp)) {
+ break;
+ }
+ }
+ }
+
+ /* At this point we want to be sitting on the first byte of the raw audio data. */
+ if (drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData) == DRWAV_FALSE) {
+ drwav_free(pWav->pMetadata, &pWav->allocationCallbacks);
+ return DRWAV_FALSE;
+ }
+
+
+ pWav->fmt = fmt;
+ pWav->sampleRate = fmt.sampleRate;
+ pWav->channels = fmt.channels;
+ pWav->bitsPerSample = fmt.bitsPerSample;
+ pWav->translatedFormatTag = translatedFormatTag;
+
+ /*
+ I've had a report where files would start glitching after seeking. The reason for this is the data
+ chunk is not a clean multiple of the PCM frame size in bytes. Where this becomes a problem is when
+ seeking, because the number of bytes remaining in the data chunk is used to calculate the current
+ byte position. If this byte position is not aligned to the number of bytes in a PCM frame, it will
+ result in the seek not being cleanly positioned at the start of the PCM frame thereby resulting in
+ all decoded frames after that being corrupted.
+
+ To address this, we need to round the data chunk size down to the nearest multiple of the frame size.
+ */
+ if (!drwav__is_compressed_format_tag(translatedFormatTag)) {
+ drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame > 0) {
+ dataChunkSize -= (dataChunkSize % bytesPerFrame);
+ }
+ }
+
+ pWav->bytesRemaining = dataChunkSize;
+ pWav->dataChunkDataSize = dataChunkSize;
+
+ if (sampleCountFromFactChunk != 0) {
+ pWav->totalPCMFrameCount = sampleCountFromFactChunk;
+ } else if (aiffFrameCount != 0) {
+ pWav->totalPCMFrameCount = aiffFrameCount;
+ } else {
+ drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ drwav_free(pWav->pMetadata, &pWav->allocationCallbacks);
+ return DRWAV_FALSE; /* Invalid file. */
+ }
+
+ pWav->totalPCMFrameCount = dataChunkSize / bytesPerFrame;
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
+ drwav_uint64 totalBlockHeaderSizeInBytes;
+ drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
+
+ /* Make sure any trailing partial block is accounted for. */
+ if ((blockCount * fmt.blockAlign) < dataChunkSize) {
+ blockCount += 1;
+ }
+
+ /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */
+ totalBlockHeaderSizeInBytes = blockCount * (6*fmt.channels);
+ pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels;
+ }
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ drwav_uint64 totalBlockHeaderSizeInBytes;
+ drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
+
+ /* Make sure any trailing partial block is accounted for. */
+ if ((blockCount * fmt.blockAlign) < dataChunkSize) {
+ blockCount += 1;
+ }
+
+ /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */
+ totalBlockHeaderSizeInBytes = blockCount * (4*fmt.channels);
+ pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels;
+
+ /* The header includes a decoded sample for each channel which acts as the initial predictor sample. */
+ pWav->totalPCMFrameCount += blockCount;
+ }
+ }
+
+ /* Some formats only support a certain number of channels. */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ if (pWav->channels > 2) {
+ drwav_free(pWav->pMetadata, &pWav->allocationCallbacks);
+ return DRWAV_FALSE;
+ }
+ }
+
+ /* The number of bytes per frame must be known. If not, it's an invalid file and not decodable. */
+ if (drwav_get_bytes_per_pcm_frame(pWav) == 0) {
+ drwav_free(pWav->pMetadata, &pWav->allocationCallbacks);
+ return DRWAV_FALSE;
+ }
+
+#ifdef DR_WAV_LIBSNDFILE_COMPAT
+ /*
+ I use libsndfile as a benchmark for testing, however in the version I'm using (from the Windows installer on the libsndfile website),
+ it appears the total sample count libsndfile uses for MS-ADPCM is incorrect. It would seem they are computing the total sample count
+ from the number of blocks, however this results in the inclusion of extra silent samples at the end of the last block. The correct
+ way to know the total sample count is to inspect the "fact" chunk, which should always be present for compressed formats, and should
+ always include the sample count. This little block of code below is only used to emulate the libsndfile logic so I can properly run my
+ correctness tests against libsndfile, and is disabled by default.
+ */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
+ drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
+ pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels; /* x2 because two samples per byte. */
+ }
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
+ pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels;
+ }
+#endif
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_ex(pWav, onRead, onSeek, onTell, NULL, pUserData, NULL, 0, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, drwav_chunk_proc onChunk, void* pReadSeekTellUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (!drwav_preinit(pWav, onRead, onSeek, onTell, pReadSeekTellUserData, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init__internal(pWav, onChunk, pChunkUserData, flags);
+}
+
+DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (!drwav_preinit(pWav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init__internal(pWav, NULL, NULL, flags | DRWAV_WITH_METADATA);
+}
+
+DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav)
+{
+ drwav_metadata *result = pWav->pMetadata;
+
+ pWav->pMetadata = NULL;
+ pWav->metadataCount = 0;
+
+ return result;
+}
+
+
+DRWAV_PRIVATE size_t drwav__write(drwav* pWav, const void* pData, size_t dataSize)
+{
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->onWrite != NULL);
+
+ /* Generic write. Assumes no byte reordering required. */
+ return pWav->onWrite(pWav->pUserData, pData, dataSize);
+}
+
+DRWAV_PRIVATE size_t drwav__write_byte(drwav* pWav, drwav_uint8 byte)
+{
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->onWrite != NULL);
+
+ return pWav->onWrite(pWav->pUserData, &byte, 1);
+}
+
+DRWAV_PRIVATE size_t drwav__write_u16ne_to_le(drwav* pWav, drwav_uint16 value)
+{
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->onWrite != NULL);
+
+ if (!drwav__is_little_endian()) {
+ value = drwav__bswap16(value);
+ }
+
+ return drwav__write(pWav, &value, 2);
+}
+
+DRWAV_PRIVATE size_t drwav__write_u32ne_to_le(drwav* pWav, drwav_uint32 value)
+{
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->onWrite != NULL);
+
+ if (!drwav__is_little_endian()) {
+ value = drwav__bswap32(value);
+ }
+
+ return drwav__write(pWav, &value, 4);
+}
+
+DRWAV_PRIVATE size_t drwav__write_u64ne_to_le(drwav* pWav, drwav_uint64 value)
+{
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->onWrite != NULL);
+
+ if (!drwav__is_little_endian()) {
+ value = drwav__bswap64(value);
+ }
+
+ return drwav__write(pWav, &value, 8);
+}
+
+DRWAV_PRIVATE size_t drwav__write_f32ne_to_le(drwav* pWav, float value)
+{
+ union {
+ drwav_uint32 u32;
+ float f32;
+ } u;
+
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->onWrite != NULL);
+
+ u.f32 = value;
+
+ if (!drwav__is_little_endian()) {
+ u.u32 = drwav__bswap32(u.u32);
+ }
+
+ return drwav__write(pWav, &u.u32, 4);
+}
+
+DRWAV_PRIVATE size_t drwav__write_or_count(drwav* pWav, const void* pData, size_t dataSize)
+{
+ if (pWav == NULL) {
+ return dataSize;
+ }
+
+ return drwav__write(pWav, pData, dataSize);
+}
+
+DRWAV_PRIVATE size_t drwav__write_or_count_byte(drwav* pWav, drwav_uint8 byte)
+{
+ if (pWav == NULL) {
+ return 1;
+ }
+
+ return drwav__write_byte(pWav, byte);
+}
+
+DRWAV_PRIVATE size_t drwav__write_or_count_u16ne_to_le(drwav* pWav, drwav_uint16 value)
+{
+ if (pWav == NULL) {
+ return 2;
+ }
+
+ return drwav__write_u16ne_to_le(pWav, value);
+}
+
+DRWAV_PRIVATE size_t drwav__write_or_count_u32ne_to_le(drwav* pWav, drwav_uint32 value)
+{
+ if (pWav == NULL) {
+ return 4;
+ }
+
+ return drwav__write_u32ne_to_le(pWav, value);
+}
+
+#if 0 /* Unused for now. */
+DRWAV_PRIVATE size_t drwav__write_or_count_u64ne_to_le(drwav* pWav, drwav_uint64 value)
+{
+ if (pWav == NULL) {
+ return 8;
+ }
+
+ return drwav__write_u64ne_to_le(pWav, value);
+}
+#endif
+
+DRWAV_PRIVATE size_t drwav__write_or_count_f32ne_to_le(drwav* pWav, float value)
+{
+ if (pWav == NULL) {
+ return 4;
+ }
+
+ return drwav__write_f32ne_to_le(pWav, value);
+}
+
+DRWAV_PRIVATE size_t drwav__write_or_count_string_to_fixed_size_buf(drwav* pWav, char* str, size_t bufFixedSize)
+{
+ size_t len;
+
+ if (pWav == NULL) {
+ return bufFixedSize;
+ }
+
+ len = drwav__strlen_clamped(str, bufFixedSize);
+ drwav__write_or_count(pWav, str, len);
+
+ if (len < bufFixedSize) {
+ size_t i;
+ for (i = 0; i < bufFixedSize - len; ++i) {
+ drwav__write_byte(pWav, 0);
+ }
+ }
+
+ return bufFixedSize;
+}
+
+
+/* pWav can be NULL meaning just count the bytes that would be written. */
+DRWAV_PRIVATE size_t drwav__write_or_count_metadata(drwav* pWav, drwav_metadata* pMetadatas, drwav_uint32 metadataCount)
+{
+ size_t bytesWritten = 0;
+ drwav_bool32 hasListAdtl = DRWAV_FALSE;
+ drwav_bool32 hasListInfo = DRWAV_FALSE;
+ drwav_uint32 iMetadata;
+
+ if (pMetadatas == NULL || metadataCount == 0) {
+ return 0;
+ }
+
+ for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) {
+ drwav_metadata* pMetadata = &pMetadatas[iMetadata];
+ drwav_uint32 chunkSize = 0;
+
+ if ((pMetadata->type & drwav_metadata_type_list_all_info_strings) || (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list)) {
+ hasListInfo = DRWAV_TRUE;
+ }
+
+ if ((pMetadata->type & drwav_metadata_type_list_all_adtl) || (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list)) {
+ hasListAdtl = DRWAV_TRUE;
+ }
+
+ switch (pMetadata->type) {
+ case drwav_metadata_type_smpl:
+ {
+ drwav_uint32 iLoop;
+
+ chunkSize = DRWAV_SMPL_BYTES + DRWAV_SMPL_LOOP_BYTES * pMetadata->data.smpl.sampleLoopCount + pMetadata->data.smpl.samplerSpecificDataSizeInBytes;
+
+ bytesWritten += drwav__write_or_count(pWav, "smpl", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.manufacturerId);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.productId);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.samplePeriodNanoseconds);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.midiUnityNote);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.midiPitchFraction);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.smpteFormat);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.smpteOffset);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.sampleLoopCount);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.samplerSpecificDataSizeInBytes);
+
+ for (iLoop = 0; iLoop < pMetadata->data.smpl.sampleLoopCount; ++iLoop) {
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].cuePointId);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].type);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].firstSampleOffset);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].lastSampleOffset);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].sampleFraction);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].playCount);
+ }
+
+ if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) {
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes);
+ }
+ } break;
+
+ case drwav_metadata_type_inst:
+ {
+ chunkSize = DRWAV_INST_BYTES;
+
+ bytesWritten += drwav__write_or_count(pWav, "inst", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.midiUnityNote, 1);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.fineTuneCents, 1);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.gainDecibels, 1);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.lowNote, 1);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.highNote, 1);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.lowVelocity, 1);
+ bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.highVelocity, 1);
+ } break;
+
+ case drwav_metadata_type_cue:
+ {
+ drwav_uint32 iCuePoint;
+
+ chunkSize = DRWAV_CUE_BYTES + DRWAV_CUE_POINT_BYTES * pMetadata->data.cue.cuePointCount;
+
+ bytesWritten += drwav__write_or_count(pWav, "cue ", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.cuePointCount);
+ for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) {
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].id);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId, 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].blockStart);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].sampleOffset);
+ }
+ } break;
+
+ case drwav_metadata_type_acid:
+ {
+ chunkSize = DRWAV_ACID_BYTES;
+
+ bytesWritten += drwav__write_or_count(pWav, "acid", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.acid.flags);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.midiUnityNote);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.reserved1);
+ bytesWritten += drwav__write_or_count_f32ne_to_le(pWav, pMetadata->data.acid.reserved2);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.acid.numBeats);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.meterDenominator);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.meterNumerator);
+ bytesWritten += drwav__write_or_count_f32ne_to_le(pWav, pMetadata->data.acid.tempo);
+ } break;
+
+ case drwav_metadata_type_bext:
+ {
+ char reservedBuf[DRWAV_BEXT_RESERVED_BYTES];
+ drwav_uint32 timeReferenceLow;
+ drwav_uint32 timeReferenceHigh;
+
+ chunkSize = DRWAV_BEXT_BYTES + pMetadata->data.bext.codingHistorySize;
+
+ bytesWritten += drwav__write_or_count(pWav, "bext", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+
+ bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pDescription, DRWAV_BEXT_DESCRIPTION_BYTES);
+ bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pOriginatorName, DRWAV_BEXT_ORIGINATOR_NAME_BYTES);
+ bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pOriginatorReference, DRWAV_BEXT_ORIGINATOR_REF_BYTES);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate));
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime));
+
+ timeReferenceLow = (drwav_uint32)(pMetadata->data.bext.timeReference & 0xFFFFFFFF);
+ timeReferenceHigh = (drwav_uint32)(pMetadata->data.bext.timeReference >> 32);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, timeReferenceLow);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, timeReferenceHigh);
+
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.version);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.loudnessValue);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.loudnessRange);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxTruePeakLevel);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxMomentaryLoudness);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxShortTermLoudness);
+
+ DRWAV_ZERO_MEMORY(reservedBuf, sizeof(reservedBuf));
+ bytesWritten += drwav__write_or_count(pWav, reservedBuf, sizeof(reservedBuf));
+
+ if (pMetadata->data.bext.codingHistorySize > 0) {
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pCodingHistory, pMetadata->data.bext.codingHistorySize);
+ }
+ } break;
+
+ case drwav_metadata_type_unknown:
+ {
+ if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_top_level) {
+ chunkSize = pMetadata->data.unknown.dataSizeInBytes;
+
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, pMetadata->data.unknown.dataSizeInBytes);
+ }
+ } break;
+
+ default: break;
+ }
+ if ((chunkSize % 2) != 0) {
+ bytesWritten += drwav__write_or_count_byte(pWav, 0);
+ }
+ }
+
+ if (hasListInfo) {
+ drwav_uint32 chunkSize = 4; /* Start with 4 bytes for "INFO". */
+ for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) {
+ drwav_metadata* pMetadata = &pMetadatas[iMetadata];
+
+ if ((pMetadata->type & drwav_metadata_type_list_all_info_strings)) {
+ chunkSize += 8; /* For id and string size. */
+ chunkSize += pMetadata->data.infoText.stringLength + 1; /* Include null terminator. */
+ } else if (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list) {
+ chunkSize += 8; /* For id string size. */
+ chunkSize += pMetadata->data.unknown.dataSizeInBytes;
+ }
+
+ if ((chunkSize % 2) != 0) {
+ chunkSize += 1;
+ }
+ }
+
+ bytesWritten += drwav__write_or_count(pWav, "LIST", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+ bytesWritten += drwav__write_or_count(pWav, "INFO", 4);
+
+ for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) {
+ drwav_metadata* pMetadata = &pMetadatas[iMetadata];
+ drwav_uint32 subchunkSize = 0;
+
+ if (pMetadata->type & drwav_metadata_type_list_all_info_strings) {
+ const char* pID = NULL;
+
+ switch (pMetadata->type) {
+ case drwav_metadata_type_list_info_software: pID = "ISFT"; break;
+ case drwav_metadata_type_list_info_copyright: pID = "ICOP"; break;
+ case drwav_metadata_type_list_info_title: pID = "INAM"; break;
+ case drwav_metadata_type_list_info_artist: pID = "IART"; break;
+ case drwav_metadata_type_list_info_comment: pID = "ICMT"; break;
+ case drwav_metadata_type_list_info_date: pID = "ICRD"; break;
+ case drwav_metadata_type_list_info_genre: pID = "IGNR"; break;
+ case drwav_metadata_type_list_info_album: pID = "IPRD"; break;
+ case drwav_metadata_type_list_info_tracknumber: pID = "ITRK"; break;
+ case drwav_metadata_type_list_info_location: pID = "IARL"; break;
+ case drwav_metadata_type_list_info_organization: pID = "ICMS"; break;
+ case drwav_metadata_type_list_info_keywords: pID = "IKEY"; break;
+ case drwav_metadata_type_list_info_medium: pID = "IMED"; break;
+ case drwav_metadata_type_list_info_description: pID = "ISBJ"; break;
+ default: break;
+ }
+
+ DRWAV_ASSERT(pID != NULL);
+
+ if (pMetadata->data.infoText.stringLength) {
+ subchunkSize = pMetadata->data.infoText.stringLength + 1;
+ bytesWritten += drwav__write_or_count(pWav, pID, 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.infoText.pString, pMetadata->data.infoText.stringLength);
+ bytesWritten += drwav__write_or_count_byte(pWav, '\0');
+ }
+ } else if (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list) {
+ if (pMetadata->data.unknown.dataSizeInBytes) {
+ subchunkSize = pMetadata->data.unknown.dataSizeInBytes;
+
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.unknown.dataSizeInBytes);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, subchunkSize);
+ }
+ }
+
+ if ((subchunkSize % 2) != 0) {
+ bytesWritten += drwav__write_or_count_byte(pWav, 0);
+ }
+ }
+ }
+
+ if (hasListAdtl) {
+ drwav_uint32 chunkSize = 4; /* start with 4 bytes for "adtl" */
+
+ for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) {
+ drwav_metadata* pMetadata = &pMetadatas[iMetadata];
+
+ switch (pMetadata->type)
+ {
+ case drwav_metadata_type_list_label:
+ case drwav_metadata_type_list_note:
+ {
+ chunkSize += 8; /* for id and chunk size */
+ chunkSize += DRWAV_LIST_LABEL_OR_NOTE_BYTES;
+
+ if (pMetadata->data.labelOrNote.stringLength > 0) {
+ chunkSize += pMetadata->data.labelOrNote.stringLength + 1;
+ }
+ } break;
+
+ case drwav_metadata_type_list_labelled_cue_region:
+ {
+ chunkSize += 8; /* for id and chunk size */
+ chunkSize += DRWAV_LIST_LABELLED_TEXT_BYTES;
+
+ if (pMetadata->data.labelledCueRegion.stringLength > 0) {
+ chunkSize += pMetadata->data.labelledCueRegion.stringLength + 1;
+ }
+ } break;
+
+ case drwav_metadata_type_unknown:
+ {
+ if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list) {
+ chunkSize += 8; /* for id and chunk size */
+ chunkSize += pMetadata->data.unknown.dataSizeInBytes;
+ }
+ } break;
+
+ default: break;
+ }
+
+ if ((chunkSize % 2) != 0) {
+ chunkSize += 1;
+ }
+ }
+
+ bytesWritten += drwav__write_or_count(pWav, "LIST", 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize);
+ bytesWritten += drwav__write_or_count(pWav, "adtl", 4);
+
+ for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) {
+ drwav_metadata* pMetadata = &pMetadatas[iMetadata];
+ drwav_uint32 subchunkSize = 0;
+
+ switch (pMetadata->type)
+ {
+ case drwav_metadata_type_list_label:
+ case drwav_metadata_type_list_note:
+ {
+ if (pMetadata->data.labelOrNote.stringLength > 0) {
+ const char *pID = NULL;
+
+ if (pMetadata->type == drwav_metadata_type_list_label) {
+ pID = "labl";
+ }
+ else if (pMetadata->type == drwav_metadata_type_list_note) {
+ pID = "note";
+ }
+
+ DRWAV_ASSERT(pID != NULL);
+ DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL);
+
+ subchunkSize = DRWAV_LIST_LABEL_OR_NOTE_BYTES;
+
+ bytesWritten += drwav__write_or_count(pWav, pID, 4);
+ subchunkSize += pMetadata->data.labelOrNote.stringLength + 1;
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize);
+
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelOrNote.cuePointId);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelOrNote.pString, pMetadata->data.labelOrNote.stringLength);
+ bytesWritten += drwav__write_or_count_byte(pWav, '\0');
+ }
+ } break;
+
+ case drwav_metadata_type_list_labelled_cue_region:
+ {
+ subchunkSize = DRWAV_LIST_LABELLED_TEXT_BYTES;
+
+ bytesWritten += drwav__write_or_count(pWav, "ltxt", 4);
+ if (pMetadata->data.labelledCueRegion.stringLength > 0) {
+ subchunkSize += pMetadata->data.labelledCueRegion.stringLength + 1;
+ }
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelledCueRegion.cuePointId);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelledCueRegion.sampleLength);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelledCueRegion.purposeId, 4);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.country);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.language);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.dialect);
+ bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.codePage);
+
+ if (pMetadata->data.labelledCueRegion.stringLength > 0) {
+ DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL);
+
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelledCueRegion.pString, pMetadata->data.labelledCueRegion.stringLength);
+ bytesWritten += drwav__write_or_count_byte(pWav, '\0');
+ }
+ } break;
+
+ case drwav_metadata_type_unknown:
+ {
+ if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list) {
+ subchunkSize = pMetadata->data.unknown.dataSizeInBytes;
+
+ DRWAV_ASSERT(pMetadata->data.unknown.pData != NULL);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4);
+ bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize);
+ bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, subchunkSize);
+ }
+ } break;
+
+ default: break;
+ }
+
+ if ((subchunkSize % 2) != 0) {
+ bytesWritten += drwav__write_or_count_byte(pWav, 0);
+ }
+ }
+ }
+
+ DRWAV_ASSERT((bytesWritten % 2) == 0);
+
+ return bytesWritten;
+}
+
+DRWAV_PRIVATE drwav_uint32 drwav__riff_chunk_size_riff(drwav_uint64 dataChunkSize, drwav_metadata* pMetadata, drwav_uint32 metadataCount)
+{
+ drwav_uint64 chunkSize = 4 + 24 + (drwav_uint64)drwav__write_or_count_metadata(NULL, pMetadata, metadataCount) + 8 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 24 = "fmt " chunk. 8 = "data" + u32 data size. */
+ if (chunkSize > 0xFFFFFFFFUL) {
+ chunkSize = 0xFFFFFFFFUL;
+ }
+
+ return (drwav_uint32)chunkSize; /* Safe cast due to the clamp above. */
+}
+
+DRWAV_PRIVATE drwav_uint32 drwav__data_chunk_size_riff(drwav_uint64 dataChunkSize)
+{
+ if (dataChunkSize <= 0xFFFFFFFFUL) {
+ return (drwav_uint32)dataChunkSize;
+ } else {
+ return 0xFFFFFFFFUL;
+ }
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__riff_chunk_size_w64(drwav_uint64 dataChunkSize)
+{
+ drwav_uint64 dataSubchunkPaddingSize = drwav__chunk_padding_size_w64(dataChunkSize);
+
+ return 80 + 24 + dataChunkSize + dataSubchunkPaddingSize; /* +24 because W64 includes the size of the GUID and size fields. */
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__data_chunk_size_w64(drwav_uint64 dataChunkSize)
+{
+ return 24 + dataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__riff_chunk_size_rf64(drwav_uint64 dataChunkSize, drwav_metadata *metadata, drwav_uint32 numMetadata)
+{
+ drwav_uint64 chunkSize = 4 + 36 + 24 + (drwav_uint64)drwav__write_or_count_metadata(NULL, metadata, numMetadata) + 8 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 36 = "ds64" chunk. 24 = "fmt " chunk. 8 = "data" + u32 data size. */
+ if (chunkSize > 0xFFFFFFFFUL) {
+ chunkSize = 0xFFFFFFFFUL;
+ }
+
+ return chunkSize;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav__data_chunk_size_rf64(drwav_uint64 dataChunkSize)
+{
+ return dataChunkSize;
+}
+
+
+
+DRWAV_PRIVATE drwav_bool32 drwav_preinit_write(drwav* pWav, const drwav_data_format* pFormat, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pWav == NULL || onWrite == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ if (!isSequential && onSeek == NULL) {
+ return DRWAV_FALSE; /* <-- onSeek is required when in non-sequential mode. */
+ }
+
+ /* Not currently supporting compressed formats. Will need to add support for the "fact" chunk before we enable this. */
+ if (pFormat->format == DR_WAVE_FORMAT_EXTENSIBLE) {
+ return DRWAV_FALSE;
+ }
+ if (pFormat->format == DR_WAVE_FORMAT_ADPCM || pFormat->format == DR_WAVE_FORMAT_DVI_ADPCM) {
+ return DRWAV_FALSE;
+ }
+
+ DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav));
+ pWav->onWrite = onWrite;
+ pWav->onSeek = onSeek;
+ pWav->pUserData = pUserData;
+ pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks);
+
+ if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) {
+ return DRWAV_FALSE; /* Invalid allocation callbacks. */
+ }
+
+ pWav->fmt.formatTag = (drwav_uint16)pFormat->format;
+ pWav->fmt.channels = (drwav_uint16)pFormat->channels;
+ pWav->fmt.sampleRate = pFormat->sampleRate;
+ pWav->fmt.avgBytesPerSec = (drwav_uint32)((pFormat->bitsPerSample * pFormat->sampleRate * pFormat->channels) / 8);
+ pWav->fmt.blockAlign = (drwav_uint16)((pFormat->channels * pFormat->bitsPerSample) / 8);
+ pWav->fmt.bitsPerSample = (drwav_uint16)pFormat->bitsPerSample;
+ pWav->fmt.extendedSize = 0;
+ pWav->isSequentialWrite = isSequential;
+
+ return DRWAV_TRUE;
+}
+
+
+DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount)
+{
+ /* The function assumes drwav_preinit_write() was called beforehand. */
+
+ size_t runningPos = 0;
+ drwav_uint64 initialDataChunkSize = 0;
+ drwav_uint64 chunkSizeFMT;
+
+ /*
+ The initial values for the "RIFF" and "data" chunks depends on whether or not we are initializing in sequential mode or not. In
+ sequential mode we set this to its final values straight away since they can be calculated from the total sample count. In non-
+ sequential mode we initialize it all to zero and fill it out in drwav_uninit() using a backwards seek.
+ */
+ if (pWav->isSequentialWrite) {
+ initialDataChunkSize = (totalSampleCount * pWav->fmt.bitsPerSample) / 8;
+
+ /*
+ The RIFF container has a limit on the number of samples. drwav is not allowing this. There's no practical limits for Wave64
+ so for the sake of simplicity I'm not doing any validation for that.
+ */
+ if (pFormat->container == drwav_container_riff) {
+ if (initialDataChunkSize > (0xFFFFFFFFUL - 36)) {
+ return DRWAV_FALSE; /* Not enough room to store every sample. */
+ }
+ }
+ }
+
+ pWav->dataChunkDataSizeTargetWrite = initialDataChunkSize;
+
+
+ /* "RIFF" chunk. */
+ if (pFormat->container == drwav_container_riff) {
+ drwav_uint32 chunkSizeRIFF = 36 + (drwav_uint32)initialDataChunkSize; /* +36 = "WAVE" + [sizeof "fmt " chunk] + [data chunk header] */
+ runningPos += drwav__write(pWav, "RIFF", 4);
+ runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeRIFF);
+ runningPos += drwav__write(pWav, "WAVE", 4);
+ } else if (pFormat->container == drwav_container_w64) {
+ drwav_uint64 chunkSizeRIFF = 80 + 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */
+ runningPos += drwav__write(pWav, drwavGUID_W64_RIFF, 16);
+ runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeRIFF);
+ runningPos += drwav__write(pWav, drwavGUID_W64_WAVE, 16);
+ } else if (pFormat->container == drwav_container_rf64) {
+ runningPos += drwav__write(pWav, "RF64", 4);
+ runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always 0xFFFFFFFF for RF64. Set to a proper value in the "ds64" chunk. */
+ runningPos += drwav__write(pWav, "WAVE", 4);
+ } else {
+ return DRWAV_FALSE; /* Container not supported for writing. */
+ }
+
+
+ /* "ds64" chunk (RF64 only). */
+ if (pFormat->container == drwav_container_rf64) {
+ drwav_uint32 initialds64ChunkSize = 28; /* 28 = [Size of RIFF (8 bytes)] + [Size of DATA (8 bytes)] + [Sample Count (8 bytes)] + [Table Length (4 bytes)]. Table length always set to 0. */
+ drwav_uint64 initialRiffChunkSize = 8 + initialds64ChunkSize + initialDataChunkSize; /* +8 for the ds64 header. */
+
+ runningPos += drwav__write(pWav, "ds64", 4);
+ runningPos += drwav__write_u32ne_to_le(pWav, initialds64ChunkSize); /* Size of ds64. */
+ runningPos += drwav__write_u64ne_to_le(pWav, initialRiffChunkSize); /* Size of RIFF. Set to true value at the end. */
+ runningPos += drwav__write_u64ne_to_le(pWav, initialDataChunkSize); /* Size of DATA. Set to true value at the end. */
+ runningPos += drwav__write_u64ne_to_le(pWav, totalSampleCount); /* Sample count. */
+ runningPos += drwav__write_u32ne_to_le(pWav, 0); /* Table length. Always set to zero in our case since we're not doing any other chunks than "DATA". */
+ }
+
+
+ /* "fmt " chunk. */
+ if (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64) {
+ chunkSizeFMT = 16;
+ runningPos += drwav__write(pWav, "fmt ", 4);
+ runningPos += drwav__write_u32ne_to_le(pWav, (drwav_uint32)chunkSizeFMT);
+ } else if (pFormat->container == drwav_container_w64) {
+ chunkSizeFMT = 40;
+ runningPos += drwav__write(pWav, drwavGUID_W64_FMT, 16);
+ runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeFMT);
+ }
+
+ runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.formatTag);
+ runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.channels);
+ runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.sampleRate);
+ runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.avgBytesPerSec);
+ runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.blockAlign);
+ runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.bitsPerSample);
+
+ /* TODO: is a 'fact' chunk required for DR_WAVE_FORMAT_IEEE_FLOAT? */
+
+ if (!pWav->isSequentialWrite && pWav->pMetadata != NULL && pWav->metadataCount > 0 && (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64)) {
+ runningPos += drwav__write_or_count_metadata(pWav, pWav->pMetadata, pWav->metadataCount);
+ }
+
+ pWav->dataChunkDataPos = runningPos;
+
+ /* "data" chunk. */
+ if (pFormat->container == drwav_container_riff) {
+ drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize;
+ runningPos += drwav__write(pWav, "data", 4);
+ runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeDATA);
+ } else if (pFormat->container == drwav_container_w64) {
+ drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */
+ runningPos += drwav__write(pWav, drwavGUID_W64_DATA, 16);
+ runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeDATA);
+ } else if (pFormat->container == drwav_container_rf64) {
+ runningPos += drwav__write(pWav, "data", 4);
+ runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always set to 0xFFFFFFFF for RF64. The true size of the data chunk is specified in the ds64 chunk. */
+ }
+
+ /* Set some properties for the client's convenience. */
+ pWav->container = pFormat->container;
+ pWav->channels = (drwav_uint16)pFormat->channels;
+ pWav->sampleRate = pFormat->sampleRate;
+ pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample;
+ pWav->translatedFormatTag = (drwav_uint16)pFormat->format;
+ pWav->dataChunkDataPos = runningPos;
+
+ return DRWAV_TRUE;
+}
+
+
+DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init_write__internal(pWav, pFormat, 0); /* DRWAV_FALSE = Not Sequential */
+}
+
+DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (!drwav_preinit_write(pWav, pFormat, DRWAV_TRUE, onWrite, NULL, pUserData, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init_write__internal(pWav, pFormat, totalSampleCount); /* DRWAV_TRUE = Sequential */
+}
+
+DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFormat == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init_write_sequential(pWav, pFormat, totalPCMFrameCount*pFormat->channels, onWrite, pUserData, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount)
+{
+ if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ pWav->pMetadata = pMetadata;
+ pWav->metadataCount = metadataCount;
+
+ return drwav_init_write__internal(pWav, pFormat, 0);
+}
+
+
+DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount)
+{
+ /* Casting totalFrameCount to drwav_int64 for VC6 compatibility. No issues in practice because nobody is going to exhaust the whole 63 bits. */
+ drwav_uint64 targetDataSizeBytes = (drwav_uint64)((drwav_int64)totalFrameCount * pFormat->channels * pFormat->bitsPerSample/8.0);
+ drwav_uint64 riffChunkSizeBytes;
+ drwav_uint64 fileSizeBytes = 0;
+
+ if (pFormat->container == drwav_container_riff) {
+ riffChunkSizeBytes = drwav__riff_chunk_size_riff(targetDataSizeBytes, pMetadata, metadataCount);
+ fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */
+ } else if (pFormat->container == drwav_container_w64) {
+ riffChunkSizeBytes = drwav__riff_chunk_size_w64(targetDataSizeBytes);
+ fileSizeBytes = riffChunkSizeBytes;
+ } else if (pFormat->container == drwav_container_rf64) {
+ riffChunkSizeBytes = drwav__riff_chunk_size_rf64(targetDataSizeBytes, pMetadata, metadataCount);
+ fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */
+ }
+
+ return fileSizeBytes;
+}
+
+
+#ifndef DR_WAV_NO_STDIO
+
+/* Errno */
+/* drwav_result_from_errno() is only used for fopen() and wfopen() so putting it inside DR_WAV_NO_STDIO for now. If something else needs this later we can move it out. */
+#include <errno.h>
+DRWAV_PRIVATE drwav_result drwav_result_from_errno(int e)
+{
+ switch (e)
+ {
+ case 0: return DRWAV_SUCCESS;
+ #ifdef EPERM
+ case EPERM: return DRWAV_INVALID_OPERATION;
+ #endif
+ #ifdef ENOENT
+ case ENOENT: return DRWAV_DOES_NOT_EXIST;
+ #endif
+ #ifdef ESRCH
+ case ESRCH: return DRWAV_DOES_NOT_EXIST;
+ #endif
+ #ifdef EINTR
+ case EINTR: return DRWAV_INTERRUPT;
+ #endif
+ #ifdef EIO
+ case EIO: return DRWAV_IO_ERROR;
+ #endif
+ #ifdef ENXIO
+ case ENXIO: return DRWAV_DOES_NOT_EXIST;
+ #endif
+ #ifdef E2BIG
+ case E2BIG: return DRWAV_INVALID_ARGS;
+ #endif
+ #ifdef ENOEXEC
+ case ENOEXEC: return DRWAV_INVALID_FILE;
+ #endif
+ #ifdef EBADF
+ case EBADF: return DRWAV_INVALID_FILE;
+ #endif
+ #ifdef ECHILD
+ case ECHILD: return DRWAV_ERROR;
+ #endif
+ #ifdef EAGAIN
+ case EAGAIN: return DRWAV_UNAVAILABLE;
+ #endif
+ #ifdef ENOMEM
+ case ENOMEM: return DRWAV_OUT_OF_MEMORY;
+ #endif
+ #ifdef EACCES
+ case EACCES: return DRWAV_ACCESS_DENIED;
+ #endif
+ #ifdef EFAULT
+ case EFAULT: return DRWAV_BAD_ADDRESS;
+ #endif
+ #ifdef ENOTBLK
+ case ENOTBLK: return DRWAV_ERROR;
+ #endif
+ #ifdef EBUSY
+ case EBUSY: return DRWAV_BUSY;
+ #endif
+ #ifdef EEXIST
+ case EEXIST: return DRWAV_ALREADY_EXISTS;
+ #endif
+ #ifdef EXDEV
+ case EXDEV: return DRWAV_ERROR;
+ #endif
+ #ifdef ENODEV
+ case ENODEV: return DRWAV_DOES_NOT_EXIST;
+ #endif
+ #ifdef ENOTDIR
+ case ENOTDIR: return DRWAV_NOT_DIRECTORY;
+ #endif
+ #ifdef EISDIR
+ case EISDIR: return DRWAV_IS_DIRECTORY;
+ #endif
+ #ifdef EINVAL
+ case EINVAL: return DRWAV_INVALID_ARGS;
+ #endif
+ #ifdef ENFILE
+ case ENFILE: return DRWAV_TOO_MANY_OPEN_FILES;
+ #endif
+ #ifdef EMFILE
+ case EMFILE: return DRWAV_TOO_MANY_OPEN_FILES;
+ #endif
+ #ifdef ENOTTY
+ case ENOTTY: return DRWAV_INVALID_OPERATION;
+ #endif
+ #ifdef ETXTBSY
+ case ETXTBSY: return DRWAV_BUSY;
+ #endif
+ #ifdef EFBIG
+ case EFBIG: return DRWAV_TOO_BIG;
+ #endif
+ #ifdef ENOSPC
+ case ENOSPC: return DRWAV_NO_SPACE;
+ #endif
+ #ifdef ESPIPE
+ case ESPIPE: return DRWAV_BAD_SEEK;
+ #endif
+ #ifdef EROFS
+ case EROFS: return DRWAV_ACCESS_DENIED;
+ #endif
+ #ifdef EMLINK
+ case EMLINK: return DRWAV_TOO_MANY_LINKS;
+ #endif
+ #ifdef EPIPE
+ case EPIPE: return DRWAV_BAD_PIPE;
+ #endif
+ #ifdef EDOM
+ case EDOM: return DRWAV_OUT_OF_RANGE;
+ #endif
+ #ifdef ERANGE
+ case ERANGE: return DRWAV_OUT_OF_RANGE;
+ #endif
+ #ifdef EDEADLK
+ case EDEADLK: return DRWAV_DEADLOCK;
+ #endif
+ #ifdef ENAMETOOLONG
+ case ENAMETOOLONG: return DRWAV_PATH_TOO_LONG;
+ #endif
+ #ifdef ENOLCK
+ case ENOLCK: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOSYS
+ case ENOSYS: return DRWAV_NOT_IMPLEMENTED;
+ #endif
+ #if defined(ENOTEMPTY) && ENOTEMPTY != EEXIST /* In AIX, ENOTEMPTY and EEXIST use the same value. */
+ case ENOTEMPTY: return DRWAV_DIRECTORY_NOT_EMPTY;
+ #endif
+ #ifdef ELOOP
+ case ELOOP: return DRWAV_TOO_MANY_LINKS;
+ #endif
+ #ifdef ENOMSG
+ case ENOMSG: return DRWAV_NO_MESSAGE;
+ #endif
+ #ifdef EIDRM
+ case EIDRM: return DRWAV_ERROR;
+ #endif
+ #ifdef ECHRNG
+ case ECHRNG: return DRWAV_ERROR;
+ #endif
+ #ifdef EL2NSYNC
+ case EL2NSYNC: return DRWAV_ERROR;
+ #endif
+ #ifdef EL3HLT
+ case EL3HLT: return DRWAV_ERROR;
+ #endif
+ #ifdef EL3RST
+ case EL3RST: return DRWAV_ERROR;
+ #endif
+ #ifdef ELNRNG
+ case ELNRNG: return DRWAV_OUT_OF_RANGE;
+ #endif
+ #ifdef EUNATCH
+ case EUNATCH: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOCSI
+ case ENOCSI: return DRWAV_ERROR;
+ #endif
+ #ifdef EL2HLT
+ case EL2HLT: return DRWAV_ERROR;
+ #endif
+ #ifdef EBADE
+ case EBADE: return DRWAV_ERROR;
+ #endif
+ #ifdef EBADR
+ case EBADR: return DRWAV_ERROR;
+ #endif
+ #ifdef EXFULL
+ case EXFULL: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOANO
+ case ENOANO: return DRWAV_ERROR;
+ #endif
+ #ifdef EBADRQC
+ case EBADRQC: return DRWAV_ERROR;
+ #endif
+ #ifdef EBADSLT
+ case EBADSLT: return DRWAV_ERROR;
+ #endif
+ #ifdef EBFONT
+ case EBFONT: return DRWAV_INVALID_FILE;
+ #endif
+ #ifdef ENOSTR
+ case ENOSTR: return DRWAV_ERROR;
+ #endif
+ #ifdef ENODATA
+ case ENODATA: return DRWAV_NO_DATA_AVAILABLE;
+ #endif
+ #ifdef ETIME
+ case ETIME: return DRWAV_TIMEOUT;
+ #endif
+ #ifdef ENOSR
+ case ENOSR: return DRWAV_NO_DATA_AVAILABLE;
+ #endif
+ #ifdef ENONET
+ case ENONET: return DRWAV_NO_NETWORK;
+ #endif
+ #ifdef ENOPKG
+ case ENOPKG: return DRWAV_ERROR;
+ #endif
+ #ifdef EREMOTE
+ case EREMOTE: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOLINK
+ case ENOLINK: return DRWAV_ERROR;
+ #endif
+ #ifdef EADV
+ case EADV: return DRWAV_ERROR;
+ #endif
+ #ifdef ESRMNT
+ case ESRMNT: return DRWAV_ERROR;
+ #endif
+ #ifdef ECOMM
+ case ECOMM: return DRWAV_ERROR;
+ #endif
+ #ifdef EPROTO
+ case EPROTO: return DRWAV_ERROR;
+ #endif
+ #ifdef EMULTIHOP
+ case EMULTIHOP: return DRWAV_ERROR;
+ #endif
+ #ifdef EDOTDOT
+ case EDOTDOT: return DRWAV_ERROR;
+ #endif
+ #ifdef EBADMSG
+ case EBADMSG: return DRWAV_BAD_MESSAGE;
+ #endif
+ #ifdef EOVERFLOW
+ case EOVERFLOW: return DRWAV_TOO_BIG;
+ #endif
+ #ifdef ENOTUNIQ
+ case ENOTUNIQ: return DRWAV_NOT_UNIQUE;
+ #endif
+ #ifdef EBADFD
+ case EBADFD: return DRWAV_ERROR;
+ #endif
+ #ifdef EREMCHG
+ case EREMCHG: return DRWAV_ERROR;
+ #endif
+ #ifdef ELIBACC
+ case ELIBACC: return DRWAV_ACCESS_DENIED;
+ #endif
+ #ifdef ELIBBAD
+ case ELIBBAD: return DRWAV_INVALID_FILE;
+ #endif
+ #ifdef ELIBSCN
+ case ELIBSCN: return DRWAV_INVALID_FILE;
+ #endif
+ #ifdef ELIBMAX
+ case ELIBMAX: return DRWAV_ERROR;
+ #endif
+ #ifdef ELIBEXEC
+ case ELIBEXEC: return DRWAV_ERROR;
+ #endif
+ #ifdef EILSEQ
+ case EILSEQ: return DRWAV_INVALID_DATA;
+ #endif
+ #ifdef ERESTART
+ case ERESTART: return DRWAV_ERROR;
+ #endif
+ #ifdef ESTRPIPE
+ case ESTRPIPE: return DRWAV_ERROR;
+ #endif
+ #ifdef EUSERS
+ case EUSERS: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOTSOCK
+ case ENOTSOCK: return DRWAV_NOT_SOCKET;
+ #endif
+ #ifdef EDESTADDRREQ
+ case EDESTADDRREQ: return DRWAV_NO_ADDRESS;
+ #endif
+ #ifdef EMSGSIZE
+ case EMSGSIZE: return DRWAV_TOO_BIG;
+ #endif
+ #ifdef EPROTOTYPE
+ case EPROTOTYPE: return DRWAV_BAD_PROTOCOL;
+ #endif
+ #ifdef ENOPROTOOPT
+ case ENOPROTOOPT: return DRWAV_PROTOCOL_UNAVAILABLE;
+ #endif
+ #ifdef EPROTONOSUPPORT
+ case EPROTONOSUPPORT: return DRWAV_PROTOCOL_NOT_SUPPORTED;
+ #endif
+ #ifdef ESOCKTNOSUPPORT
+ case ESOCKTNOSUPPORT: return DRWAV_SOCKET_NOT_SUPPORTED;
+ #endif
+ #ifdef EOPNOTSUPP
+ case EOPNOTSUPP: return DRWAV_INVALID_OPERATION;
+ #endif
+ #ifdef EPFNOSUPPORT
+ case EPFNOSUPPORT: return DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED;
+ #endif
+ #ifdef EAFNOSUPPORT
+ case EAFNOSUPPORT: return DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED;
+ #endif
+ #ifdef EADDRINUSE
+ case EADDRINUSE: return DRWAV_ALREADY_IN_USE;
+ #endif
+ #ifdef EADDRNOTAVAIL
+ case EADDRNOTAVAIL: return DRWAV_ERROR;
+ #endif
+ #ifdef ENETDOWN
+ case ENETDOWN: return DRWAV_NO_NETWORK;
+ #endif
+ #ifdef ENETUNREACH
+ case ENETUNREACH: return DRWAV_NO_NETWORK;
+ #endif
+ #ifdef ENETRESET
+ case ENETRESET: return DRWAV_NO_NETWORK;
+ #endif
+ #ifdef ECONNABORTED
+ case ECONNABORTED: return DRWAV_NO_NETWORK;
+ #endif
+ #ifdef ECONNRESET
+ case ECONNRESET: return DRWAV_CONNECTION_RESET;
+ #endif
+ #ifdef ENOBUFS
+ case ENOBUFS: return DRWAV_NO_SPACE;
+ #endif
+ #ifdef EISCONN
+ case EISCONN: return DRWAV_ALREADY_CONNECTED;
+ #endif
+ #ifdef ENOTCONN
+ case ENOTCONN: return DRWAV_NOT_CONNECTED;
+ #endif
+ #ifdef ESHUTDOWN
+ case ESHUTDOWN: return DRWAV_ERROR;
+ #endif
+ #ifdef ETOOMANYREFS
+ case ETOOMANYREFS: return DRWAV_ERROR;
+ #endif
+ #ifdef ETIMEDOUT
+ case ETIMEDOUT: return DRWAV_TIMEOUT;
+ #endif
+ #ifdef ECONNREFUSED
+ case ECONNREFUSED: return DRWAV_CONNECTION_REFUSED;
+ #endif
+ #ifdef EHOSTDOWN
+ case EHOSTDOWN: return DRWAV_NO_HOST;
+ #endif
+ #ifdef EHOSTUNREACH
+ case EHOSTUNREACH: return DRWAV_NO_HOST;
+ #endif
+ #ifdef EALREADY
+ case EALREADY: return DRWAV_IN_PROGRESS;
+ #endif
+ #ifdef EINPROGRESS
+ case EINPROGRESS: return DRWAV_IN_PROGRESS;
+ #endif
+ #ifdef ESTALE
+ case ESTALE: return DRWAV_INVALID_FILE;
+ #endif
+ #ifdef EUCLEAN
+ case EUCLEAN: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOTNAM
+ case ENOTNAM: return DRWAV_ERROR;
+ #endif
+ #ifdef ENAVAIL
+ case ENAVAIL: return DRWAV_ERROR;
+ #endif
+ #ifdef EISNAM
+ case EISNAM: return DRWAV_ERROR;
+ #endif
+ #ifdef EREMOTEIO
+ case EREMOTEIO: return DRWAV_IO_ERROR;
+ #endif
+ #ifdef EDQUOT
+ case EDQUOT: return DRWAV_NO_SPACE;
+ #endif
+ #ifdef ENOMEDIUM
+ case ENOMEDIUM: return DRWAV_DOES_NOT_EXIST;
+ #endif
+ #ifdef EMEDIUMTYPE
+ case EMEDIUMTYPE: return DRWAV_ERROR;
+ #endif
+ #ifdef ECANCELED
+ case ECANCELED: return DRWAV_CANCELLED;
+ #endif
+ #ifdef ENOKEY
+ case ENOKEY: return DRWAV_ERROR;
+ #endif
+ #ifdef EKEYEXPIRED
+ case EKEYEXPIRED: return DRWAV_ERROR;
+ #endif
+ #ifdef EKEYREVOKED
+ case EKEYREVOKED: return DRWAV_ERROR;
+ #endif
+ #ifdef EKEYREJECTED
+ case EKEYREJECTED: return DRWAV_ERROR;
+ #endif
+ #ifdef EOWNERDEAD
+ case EOWNERDEAD: return DRWAV_ERROR;
+ #endif
+ #ifdef ENOTRECOVERABLE
+ case ENOTRECOVERABLE: return DRWAV_ERROR;
+ #endif
+ #ifdef ERFKILL
+ case ERFKILL: return DRWAV_ERROR;
+ #endif
+ #ifdef EHWPOISON
+ case EHWPOISON: return DRWAV_ERROR;
+ #endif
+ default: return DRWAV_ERROR;
+ }
+}
+/* End Errno */
+
+/* fopen */
+DRWAV_PRIVATE drwav_result drwav_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode)
+{
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ errno_t err;
+#endif
+
+ if (ppFile != NULL) {
+ *ppFile = NULL; /* Safety. */
+ }
+
+ if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ err = fopen_s(ppFile, pFilePath, pOpenMode);
+ if (err != 0) {
+ return drwav_result_from_errno(err);
+ }
+#else
+#if defined(_WIN32) || defined(__APPLE__)
+ *ppFile = fopen(pFilePath, pOpenMode);
+#else
+ #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE)
+ *ppFile = fopen64(pFilePath, pOpenMode);
+ #else
+ *ppFile = fopen(pFilePath, pOpenMode);
+ #endif
+#endif
+ if (*ppFile == NULL) {
+ drwav_result result = drwav_result_from_errno(errno);
+ if (result == DRWAV_SUCCESS) {
+ result = DRWAV_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */
+ }
+
+ return result;
+ }
+#endif
+
+ return DRWAV_SUCCESS;
+}
+
+/*
+_wfopen() isn't always available in all compilation environments.
+
+ * Windows only.
+ * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back).
+ * MinGW-64 (both 32- and 64-bit) seems to support it.
+ * MinGW wraps it in !defined(__STRICT_ANSI__).
+ * OpenWatcom wraps it in !defined(_NO_EXT_KEYS).
+
+This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs()
+fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support.
+*/
+#if defined(_WIN32)
+ #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS))
+ #define DRWAV_HAS_WFOPEN
+ #endif
+#endif
+
+#ifndef DR_WAV_NO_WCHAR
+DRWAV_PRIVATE drwav_result drwav_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (ppFile != NULL) {
+ *ppFile = NULL; /* Safety. */
+ }
+
+ if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+#if defined(DRWAV_HAS_WFOPEN)
+ {
+ /* Use _wfopen() on Windows. */
+ #if defined(_MSC_VER) && _MSC_VER >= 1400
+ errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode);
+ if (err != 0) {
+ return drwav_result_from_errno(err);
+ }
+ #else
+ *ppFile = _wfopen(pFilePath, pOpenMode);
+ if (*ppFile == NULL) {
+ return drwav_result_from_errno(errno);
+ }
+ #endif
+ (void)pAllocationCallbacks;
+ }
+#else
+ /*
+ Use fopen() on anything other than Windows. Requires a conversion. This is annoying because
+ fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note
+ that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for
+ maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler
+ error I'll look into improving compatibility.
+ */
+
+ /*
+ Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just
+ need to abort with an error. If you encounter a compiler lacking such support, add it to this list
+ and submit a bug report and it'll be added to the library upstream.
+ */
+ #if defined(__DJGPP__)
+ {
+ /* Nothing to do here. This will fall through to the error check below. */
+ }
+ #else
+ {
+ mbstate_t mbs;
+ size_t lenMB;
+ const wchar_t* pFilePathTemp = pFilePath;
+ char* pFilePathMB = NULL;
+ char pOpenModeMB[32] = {0};
+
+ /* Get the length first. */
+ DRWAV_ZERO_OBJECT(&mbs);
+ lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs);
+ if (lenMB == (size_t)-1) {
+ return drwav_result_from_errno(errno);
+ }
+
+ pFilePathMB = (char*)drwav__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks);
+ if (pFilePathMB == NULL) {
+ return DRWAV_OUT_OF_MEMORY;
+ }
+
+ pFilePathTemp = pFilePath;
+ DRWAV_ZERO_OBJECT(&mbs);
+ wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs);
+
+ /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */
+ {
+ size_t i = 0;
+ for (;;) {
+ if (pOpenMode[i] == 0) {
+ pOpenModeMB[i] = '\0';
+ break;
+ }
+
+ pOpenModeMB[i] = (char)pOpenMode[i];
+ i += 1;
+ }
+ }
+
+ *ppFile = fopen(pFilePathMB, pOpenModeMB);
+
+ drwav__free_from_callbacks(pFilePathMB, pAllocationCallbacks);
+ }
+ #endif
+
+ if (*ppFile == NULL) {
+ return DRWAV_ERROR;
+ }
+#endif
+
+ return DRWAV_SUCCESS;
+}
+#endif
+/* End fopen */
+
+
+DRWAV_PRIVATE size_t drwav__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead)
+{
+ return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData);
+}
+
+DRWAV_PRIVATE size_t drwav__on_write_stdio(void* pUserData, const void* pData, size_t bytesToWrite)
+{
+ return fwrite(pData, 1, bytesToWrite, (FILE*)pUserData);
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__on_seek_stdio(void* pUserData, int offset, drwav_seek_origin origin)
+{
+ int whence = SEEK_SET;
+ if (origin == DRWAV_SEEK_CUR) {
+ whence = SEEK_CUR;
+ } else if (origin == DRWAV_SEEK_END) {
+ whence = SEEK_END;
+ }
+
+ return fseek((FILE*)pUserData, offset, whence) == 0;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__on_tell_stdio(void* pUserData, drwav_int64* pCursor)
+{
+ FILE* pFileStdio = (FILE*)pUserData;
+ drwav_int64 result;
+
+ /* These were all validated at a higher level. */
+ DRWAV_ASSERT(pFileStdio != NULL);
+ DRWAV_ASSERT(pCursor != NULL);
+
+#if defined(_WIN32) && !defined(NXDK)
+ #if defined(_MSC_VER) && _MSC_VER > 1200
+ result = _ftelli64(pFileStdio);
+ #else
+ result = ftell(pFileStdio);
+ #endif
+#else
+ result = ftell(pFileStdio);
+#endif
+
+ *pCursor = result;
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_file_ex(pWav, filename, NULL, NULL, 0, pAllocationCallbacks);
+}
+
+
+DRWAV_PRIVATE drwav_bool32 drwav_init_file__internal_FILE(drwav* pWav, FILE* pFile, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav_bool32 result;
+
+ result = drwav_preinit(pWav, drwav__on_read_stdio, drwav__on_seek_stdio, drwav__on_tell_stdio, (void*)pFile, pAllocationCallbacks);
+ if (result != DRWAV_TRUE) {
+ fclose(pFile);
+ return result;
+ }
+
+ result = drwav_init__internal(pWav, onChunk, pChunkUserData, flags);
+ if (result != DRWAV_TRUE) {
+ fclose(pFile);
+ return result;
+ }
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ FILE* pFile;
+ if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ /* This takes ownership of the FILE* object. */
+ return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, pAllocationCallbacks);
+}
+
+#ifndef DR_WAV_NO_WCHAR
+DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_file_ex_w(pWav, filename, NULL, NULL, 0, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ FILE* pFile;
+ if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ /* This takes ownership of the FILE* object. */
+ return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, pAllocationCallbacks);
+}
+#endif
+
+DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ FILE* pFile;
+ if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ /* This takes ownership of the FILE* object. */
+ return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags | DRWAV_WITH_METADATA, pAllocationCallbacks);
+}
+
+#ifndef DR_WAV_NO_WCHAR
+DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ FILE* pFile;
+ if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ /* This takes ownership of the FILE* object. */
+ return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags | DRWAV_WITH_METADATA, pAllocationCallbacks);
+}
+#endif
+
+
+DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal_FILE(drwav* pWav, FILE* pFile, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav_bool32 result;
+
+ result = drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks);
+ if (result != DRWAV_TRUE) {
+ fclose(pFile);
+ return result;
+ }
+
+ result = drwav_init_write__internal(pWav, pFormat, totalSampleCount);
+ if (result != DRWAV_TRUE) {
+ fclose(pFile);
+ return result;
+ }
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ FILE* pFile;
+ if (drwav_fopen(&pFile, filename, "wb") != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ /* This takes ownership of the FILE* object. */
+ return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks);
+}
+
+#ifndef DR_WAV_NO_WCHAR
+DRWAV_PRIVATE drwav_bool32 drwav_init_file_write_w__internal(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ FILE* pFile;
+ if (drwav_wfopen(&pFile, filename, L"wb", pAllocationCallbacks) != DRWAV_SUCCESS) {
+ return DRWAV_FALSE;
+ }
+
+ /* This takes ownership of the FILE* object. */
+ return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks);
+}
+#endif
+
+DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFormat == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init_file_write_sequential(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks);
+}
+
+#ifndef DR_WAV_NO_WCHAR
+DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_file_write_w__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_file_write_w__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFormat == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init_file_write_sequential_w(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks);
+}
+#endif
+#endif /* DR_WAV_NO_STDIO */
+
+
+DRWAV_PRIVATE size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead)
+{
+ drwav* pWav = (drwav*)pUserData;
+ size_t bytesRemaining;
+
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->memoryStream.dataSize >= pWav->memoryStream.currentReadPos);
+
+ bytesRemaining = pWav->memoryStream.dataSize - pWav->memoryStream.currentReadPos;
+ if (bytesToRead > bytesRemaining) {
+ bytesToRead = bytesRemaining;
+ }
+
+ if (bytesToRead > 0) {
+ DRWAV_COPY_MEMORY(pBufferOut, pWav->memoryStream.data + pWav->memoryStream.currentReadPos, bytesToRead);
+ pWav->memoryStream.currentReadPos += bytesToRead;
+ }
+
+ return bytesToRead;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin)
+{
+ drwav* pWav = (drwav*)pUserData;
+ drwav_int64 newCursor;
+
+ DRWAV_ASSERT(pWav != NULL);
+
+ if (origin == DRWAV_SEEK_SET) {
+ newCursor = 0;
+ } else if (origin == DRWAV_SEEK_CUR) {
+ newCursor = (drwav_int64)pWav->memoryStream.currentReadPos;
+ } else if (origin == DRWAV_SEEK_END) {
+ newCursor = (drwav_int64)pWav->memoryStream.dataSize;
+ } else {
+ DRWAV_ASSERT(!"Invalid seek origin");
+ return DRWAV_FALSE;
+ }
+
+ newCursor += offset;
+
+ if (newCursor < 0) {
+ return DRWAV_FALSE; /* Trying to seek prior to the start of the buffer. */
+ }
+ if ((size_t)newCursor > pWav->memoryStream.dataSize) {
+ return DRWAV_FALSE; /* Trying to seek beyond the end of the buffer. */
+ }
+
+ pWav->memoryStream.currentReadPos = (size_t)newCursor;
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_PRIVATE size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite)
+{
+ drwav* pWav = (drwav*)pUserData;
+ size_t bytesRemaining;
+
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pWav->memoryStreamWrite.dataCapacity >= pWav->memoryStreamWrite.currentWritePos);
+
+ bytesRemaining = pWav->memoryStreamWrite.dataCapacity - pWav->memoryStreamWrite.currentWritePos;
+ if (bytesRemaining < bytesToWrite) {
+ /* Need to reallocate. */
+ void* pNewData;
+ size_t newDataCapacity = (pWav->memoryStreamWrite.dataCapacity == 0) ? 256 : pWav->memoryStreamWrite.dataCapacity * 2;
+
+ /* If doubling wasn't enough, just make it the minimum required size to write the data. */
+ if ((newDataCapacity - pWav->memoryStreamWrite.currentWritePos) < bytesToWrite) {
+ newDataCapacity = pWav->memoryStreamWrite.currentWritePos + bytesToWrite;
+ }
+
+ pNewData = drwav__realloc_from_callbacks(*pWav->memoryStreamWrite.ppData, newDataCapacity, pWav->memoryStreamWrite.dataCapacity, &pWav->allocationCallbacks);
+ if (pNewData == NULL) {
+ return 0;
+ }
+
+ *pWav->memoryStreamWrite.ppData = pNewData;
+ pWav->memoryStreamWrite.dataCapacity = newDataCapacity;
+ }
+
+ DRWAV_COPY_MEMORY(((drwav_uint8*)(*pWav->memoryStreamWrite.ppData)) + pWav->memoryStreamWrite.currentWritePos, pDataIn, bytesToWrite);
+
+ pWav->memoryStreamWrite.currentWritePos += bytesToWrite;
+ if (pWav->memoryStreamWrite.dataSize < pWav->memoryStreamWrite.currentWritePos) {
+ pWav->memoryStreamWrite.dataSize = pWav->memoryStreamWrite.currentWritePos;
+ }
+
+ *pWav->memoryStreamWrite.pDataSize = pWav->memoryStreamWrite.dataSize;
+
+ return bytesToWrite;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory_write(void* pUserData, int offset, drwav_seek_origin origin)
+{
+ drwav* pWav = (drwav*)pUserData;
+ drwav_int64 newCursor;
+
+ DRWAV_ASSERT(pWav != NULL);
+
+ if (origin == DRWAV_SEEK_SET) {
+ newCursor = 0;
+ } else if (origin == DRWAV_SEEK_CUR) {
+ newCursor = (drwav_int64)pWav->memoryStreamWrite.currentWritePos;
+ } else if (origin == DRWAV_SEEK_END) {
+ newCursor = (drwav_int64)pWav->memoryStreamWrite.dataSize;
+ } else {
+ DRWAV_ASSERT(!"Invalid seek origin");
+ return DRWAV_FALSE;
+ }
+
+ newCursor += offset;
+
+ if (newCursor < 0) {
+ return DRWAV_FALSE; /* Trying to seek prior to the start of the buffer. */
+ }
+ if ((size_t)newCursor > pWav->memoryStreamWrite.dataSize) {
+ return DRWAV_FALSE; /* Trying to seek beyond the end of the buffer. */
+ }
+
+ pWav->memoryStreamWrite.currentWritePos = (size_t)newCursor;
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_PRIVATE drwav_bool32 drwav__on_tell_memory(void* pUserData, drwav_int64* pCursor)
+{
+ drwav* pWav = (drwav*)pUserData;
+
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(pCursor != NULL);
+
+ *pCursor = (drwav_int64)pWav->memoryStream.currentReadPos;
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_memory_ex(pWav, data, dataSize, NULL, NULL, 0, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (data == NULL || dataSize == 0) {
+ return DRWAV_FALSE;
+ }
+
+ if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, drwav__on_tell_memory, pWav, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ pWav->memoryStream.data = (const drwav_uint8*)data;
+ pWav->memoryStream.dataSize = dataSize;
+ pWav->memoryStream.currentReadPos = 0;
+
+ return drwav_init__internal(pWav, onChunk, pChunkUserData, flags);
+}
+
+DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (data == NULL || dataSize == 0) {
+ return DRWAV_FALSE;
+ }
+
+ if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, drwav__on_tell_memory, pWav, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ pWav->memoryStream.data = (const drwav_uint8*)data;
+ pWav->memoryStream.dataSize = dataSize;
+ pWav->memoryStream.currentReadPos = 0;
+
+ return drwav_init__internal(pWav, NULL, NULL, flags | DRWAV_WITH_METADATA);
+}
+
+
+DRWAV_PRIVATE drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (ppData == NULL || pDataSize == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ *ppData = NULL; /* Important because we're using realloc()! */
+ *pDataSize = 0;
+
+ if (!drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, pWav, pAllocationCallbacks)) {
+ return DRWAV_FALSE;
+ }
+
+ pWav->memoryStreamWrite.ppData = ppData;
+ pWav->memoryStreamWrite.pDataSize = pDataSize;
+ pWav->memoryStreamWrite.dataSize = 0;
+ pWav->memoryStreamWrite.dataCapacity = 0;
+ pWav->memoryStreamWrite.currentWritePos = 0;
+
+ return drwav_init_write__internal(pWav, pFormat, totalSampleCount);
+}
+
+DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks);
+}
+
+DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFormat == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ return drwav_init_memory_write_sequential(pWav, ppData, pDataSize, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks);
+}
+
+
+
+DRWAV_API drwav_result drwav_uninit(drwav* pWav)
+{
+ drwav_result result = DRWAV_SUCCESS;
+
+ if (pWav == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+ /*
+ If the drwav object was opened in write mode we'll need to finalize a few things:
+ - Make sure the "data" chunk is aligned to 16-bits for RIFF containers, or 64 bits for W64 containers.
+ - Set the size of the "data" chunk.
+ */
+ if (pWav->onWrite != NULL) {
+ drwav_uint32 paddingSize = 0;
+
+ /* Padding. Do not adjust pWav->dataChunkDataSize - this should not include the padding. */
+ if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) {
+ paddingSize = drwav__chunk_padding_size_riff(pWav->dataChunkDataSize);
+ } else {
+ paddingSize = drwav__chunk_padding_size_w64(pWav->dataChunkDataSize);
+ }
+
+ if (paddingSize > 0) {
+ drwav_uint64 paddingData = 0;
+ drwav__write(pWav, &paddingData, paddingSize); /* Byte order does not matter for this. */
+ }
+
+ /*
+ Chunk sizes. When using sequential mode, these will have been filled in at initialization time. We only need
+ to do this when using non-sequential mode.
+ */
+ if (pWav->onSeek && !pWav->isSequentialWrite) {
+ if (pWav->container == drwav_container_riff) {
+ /* The "RIFF" chunk size. */
+ if (pWav->onSeek(pWav->pUserData, 4, DRWAV_SEEK_SET)) {
+ drwav_uint32 riffChunkSize = drwav__riff_chunk_size_riff(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount);
+ drwav__write_u32ne_to_le(pWav, riffChunkSize);
+ }
+
+ /* The "data" chunk size. */
+ if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 4, DRWAV_SEEK_SET)) {
+ drwav_uint32 dataChunkSize = drwav__data_chunk_size_riff(pWav->dataChunkDataSize);
+ drwav__write_u32ne_to_le(pWav, dataChunkSize);
+ }
+ } else if (pWav->container == drwav_container_w64) {
+ /* The "RIFF" chunk size. */
+ if (pWav->onSeek(pWav->pUserData, 16, DRWAV_SEEK_SET)) {
+ drwav_uint64 riffChunkSize = drwav__riff_chunk_size_w64(pWav->dataChunkDataSize);
+ drwav__write_u64ne_to_le(pWav, riffChunkSize);
+ }
+
+ /* The "data" chunk size. */
+ if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 8, DRWAV_SEEK_SET)) {
+ drwav_uint64 dataChunkSize = drwav__data_chunk_size_w64(pWav->dataChunkDataSize);
+ drwav__write_u64ne_to_le(pWav, dataChunkSize);
+ }
+ } else if (pWav->container == drwav_container_rf64) {
+ /* We only need to update the ds64 chunk. The "RIFF" and "data" chunks always have their sizes set to 0xFFFFFFFF for RF64. */
+ int ds64BodyPos = 12 + 8;
+
+ /* The "RIFF" chunk size. */
+ if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 0, DRWAV_SEEK_SET)) {
+ drwav_uint64 riffChunkSize = drwav__riff_chunk_size_rf64(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount);
+ drwav__write_u64ne_to_le(pWav, riffChunkSize);
+ }
+
+ /* The "data" chunk size. */
+ if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 8, DRWAV_SEEK_SET)) {
+ drwav_uint64 dataChunkSize = drwav__data_chunk_size_rf64(pWav->dataChunkDataSize);
+ drwav__write_u64ne_to_le(pWav, dataChunkSize);
+ }
+ }
+ }
+
+ /* Validation for sequential mode. */
+ if (pWav->isSequentialWrite) {
+ if (pWav->dataChunkDataSize != pWav->dataChunkDataSizeTargetWrite) {
+ result = DRWAV_INVALID_FILE;
+ }
+ }
+ } else {
+ drwav_free(pWav->pMetadata, &pWav->allocationCallbacks);
+ }
+
+#ifndef DR_WAV_NO_STDIO
+ /*
+ If we opened the file with drwav_open_file() we will want to close the file handle. We can know whether or not drwav_open_file()
+ was used by looking at the onRead and onSeek callbacks.
+ */
+ if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) {
+ fclose((FILE*)pWav->pUserData);
+ }
+#endif
+
+ return result;
+}
+
+
+
+DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut)
+{
+ size_t bytesRead;
+ drwav_uint32 bytesPerFrame;
+
+ if (pWav == NULL || bytesToRead == 0) {
+ return 0; /* Invalid args. */
+ }
+
+ if (bytesToRead > pWav->bytesRemaining) {
+ bytesToRead = (size_t)pWav->bytesRemaining;
+ }
+
+ if (bytesToRead == 0) {
+ return 0; /* At end. */
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0; /* Could not determine the bytes per frame. */
+ }
+
+ if (pBufferOut != NULL) {
+ bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead);
+ } else {
+ /* We need to seek. If we fail, we need to read-and-discard to make sure we get a good byte count. */
+ bytesRead = 0;
+ while (bytesRead < bytesToRead) {
+ size_t bytesToSeek = (bytesToRead - bytesRead);
+ if (bytesToSeek > 0x7FFFFFFF) {
+ bytesToSeek = 0x7FFFFFFF;
+ }
+
+ if (pWav->onSeek(pWav->pUserData, (int)bytesToSeek, DRWAV_SEEK_CUR) == DRWAV_FALSE) {
+ break;
+ }
+
+ bytesRead += bytesToSeek;
+ }
+
+ /* When we get here we may need to read-and-discard some data. */
+ while (bytesRead < bytesToRead) {
+ drwav_uint8 buffer[4096];
+ size_t bytesSeeked;
+ size_t bytesToSeek = (bytesToRead - bytesRead);
+ if (bytesToSeek > sizeof(buffer)) {
+ bytesToSeek = sizeof(buffer);
+ }
+
+ bytesSeeked = pWav->onRead(pWav->pUserData, buffer, bytesToSeek);
+ bytesRead += bytesSeeked;
+
+ if (bytesSeeked < bytesToSeek) {
+ break; /* Reached the end. */
+ }
+ }
+ }
+
+ pWav->readCursorInPCMFrames += bytesRead / bytesPerFrame;
+
+ pWav->bytesRemaining -= bytesRead;
+ return bytesRead;
+}
+
+
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut)
+{
+ drwav_uint32 bytesPerFrame;
+ drwav_uint64 bytesToRead; /* Intentionally uint64 instead of size_t so we can do a check that we're not reading too much on 32-bit builds. */
+ drwav_uint64 framesRemainingInFile;
+
+ if (pWav == NULL || framesToRead == 0) {
+ return 0;
+ }
+
+ /* Cannot use this function for compressed formats. */
+ if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
+ return 0;
+ }
+
+ framesRemainingInFile = pWav->totalPCMFrameCount - pWav->readCursorInPCMFrames;
+ if (framesToRead > framesRemainingInFile) {
+ framesToRead = framesRemainingInFile;
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ /* Don't try to read more samples than can potentially fit in the output buffer. */
+ bytesToRead = framesToRead * bytesPerFrame;
+ if (bytesToRead > DRWAV_SIZE_MAX) {
+ bytesToRead = (DRWAV_SIZE_MAX / bytesPerFrame) * bytesPerFrame; /* Round the number of bytes to read to a clean frame boundary. */
+ }
+
+ /*
+ Doing an explicit check here just to make it clear that we don't want to be attempt to read anything if there's no bytes to read. There
+ *could* be a time where it evaluates to 0 due to overflowing.
+ */
+ if (bytesToRead == 0) {
+ return 0;
+ }
+
+ return drwav_read_raw(pWav, (size_t)bytesToRead, pBufferOut) / bytesPerFrame;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut);
+
+ if (pBufferOut != NULL) {
+ drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0; /* Could not get the bytes per frame which means bytes per sample cannot be determined and we don't know how to byte swap. */
+ }
+
+ drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, bytesPerFrame/pWav->channels);
+ }
+
+ return framesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut)
+{
+ drwav_uint64 framesRead = 0;
+
+ if (drwav_is_container_be(pWav->container)) {
+ /*
+ Special case for AIFF. AIFF is a big-endian encoded format, but it supports a format that is
+ PCM in little-endian encoding. In this case, we fall through this branch and treate it as
+ little-endian.
+ */
+ if (pWav->container != drwav_container_aiff || pWav->aiff.isLE == DRWAV_FALSE) {
+ if (drwav__is_little_endian()) {
+ framesRead = drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut);
+ } else {
+ framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut);
+ }
+
+ goto post_process;
+ }
+ }
+
+ /* Getting here means the data should be considered little-endian. */
+ if (drwav__is_little_endian()) {
+ framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut);
+ } else {
+ framesRead = drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut);
+ }
+
+ /*
+ Here is where we check if we need to do a signed/unsigned conversion for AIFF. The reason we need to do this
+ is because dr_wav always assumes an 8-bit sample is unsigned, whereas AIFF can have signed 8-bit formats.
+ */
+ post_process:
+ {
+ if (pWav->container == drwav_container_aiff && pWav->bitsPerSample == 8 && pWav->aiff.isUnsigned == DRWAV_FALSE) {
+ if (pBufferOut != NULL) {
+ drwav_uint64 iSample;
+
+ for (iSample = 0; iSample < framesRead * pWav->channels; iSample += 1) {
+ ((drwav_uint8*)pBufferOut)[iSample] += 128;
+ }
+ }
+ }
+ }
+
+ return framesRead;
+}
+
+
+
+DRWAV_PRIVATE drwav_bool32 drwav_seek_to_first_pcm_frame(drwav* pWav)
+{
+ if (pWav->onWrite != NULL) {
+ return DRWAV_FALSE; /* No seeking in write mode. */
+ }
+
+ if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, DRWAV_SEEK_SET)) {
+ return DRWAV_FALSE;
+ }
+
+ if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
+ /* Cached data needs to be cleared for compressed formats. */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
+ DRWAV_ZERO_OBJECT(&pWav->msadpcm);
+ } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ DRWAV_ZERO_OBJECT(&pWav->ima);
+ } else {
+ DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */
+ }
+ }
+
+ pWav->readCursorInPCMFrames = 0;
+ pWav->bytesRemaining = pWav->dataChunkDataSize;
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex)
+{
+ /* Seeking should be compatible with wave files > 2GB. */
+
+ if (pWav == NULL || pWav->onSeek == NULL) {
+ return DRWAV_FALSE;
+ }
+
+ /* No seeking in write mode. */
+ if (pWav->onWrite != NULL) {
+ return DRWAV_FALSE;
+ }
+
+ /* If there are no samples, just return DRWAV_TRUE without doing anything. */
+ if (pWav->totalPCMFrameCount == 0) {
+ return DRWAV_TRUE;
+ }
+
+ /* Make sure the sample is clamped. */
+ if (targetFrameIndex > pWav->totalPCMFrameCount) {
+ targetFrameIndex = pWav->totalPCMFrameCount;
+ }
+
+ /*
+ For compressed formats we just use a slow generic seek. If we are seeking forward we just seek forward. If we are going backwards we need
+ to seek back to the start.
+ */
+ if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
+ /* TODO: This can be optimized. */
+
+ /*
+ If we're seeking forward it's simple - just keep reading samples until we hit the sample we're requesting. If we're seeking backwards,
+ we first need to seek back to the start and then just do the same thing as a forward seek.
+ */
+ if (targetFrameIndex < pWav->readCursorInPCMFrames) {
+ if (!drwav_seek_to_first_pcm_frame(pWav)) {
+ return DRWAV_FALSE;
+ }
+ }
+
+ if (targetFrameIndex > pWav->readCursorInPCMFrames) {
+ drwav_uint64 offsetInFrames = targetFrameIndex - pWav->readCursorInPCMFrames;
+
+ drwav_int16 devnull[2048];
+ while (offsetInFrames > 0) {
+ drwav_uint64 framesRead = 0;
+ drwav_uint64 framesToRead = offsetInFrames;
+ if (framesToRead > drwav_countof(devnull)/pWav->channels) {
+ framesToRead = drwav_countof(devnull)/pWav->channels;
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
+ framesRead = drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, devnull);
+ } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ framesRead = drwav_read_pcm_frames_s16__ima(pWav, framesToRead, devnull);
+ } else {
+ DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */
+ }
+
+ if (framesRead != framesToRead) {
+ return DRWAV_FALSE;
+ }
+
+ offsetInFrames -= framesRead;
+ }
+ }
+ } else {
+ drwav_uint64 totalSizeInBytes;
+ drwav_uint64 currentBytePos;
+ drwav_uint64 targetBytePos;
+ drwav_uint64 offset;
+ drwav_uint32 bytesPerFrame;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return DRWAV_FALSE; /* Not able to calculate offset. */
+ }
+
+ totalSizeInBytes = pWav->totalPCMFrameCount * bytesPerFrame;
+ /*DRWAV_ASSERT(totalSizeInBytes >= pWav->bytesRemaining);*/
+
+ currentBytePos = totalSizeInBytes - pWav->bytesRemaining;
+ targetBytePos = targetFrameIndex * bytesPerFrame;
+
+ if (currentBytePos < targetBytePos) {
+ /* Offset forwards. */
+ offset = (targetBytePos - currentBytePos);
+ } else {
+ /* Offset backwards. */
+ if (!drwav_seek_to_first_pcm_frame(pWav)) {
+ return DRWAV_FALSE;
+ }
+ offset = targetBytePos;
+ }
+
+ while (offset > 0) {
+ int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset);
+ if (!pWav->onSeek(pWav->pUserData, offset32, DRWAV_SEEK_CUR)) {
+ return DRWAV_FALSE;
+ }
+
+ pWav->readCursorInPCMFrames += offset32 / bytesPerFrame;
+ pWav->bytesRemaining -= offset32;
+ offset -= offset32;
+ }
+ }
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor)
+{
+ if (pCursor == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+ *pCursor = 0; /* Safety. */
+
+ if (pWav == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+ *pCursor = pWav->readCursorInPCMFrames;
+
+ return DRWAV_SUCCESS;
+}
+
+DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength)
+{
+ if (pLength == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+ *pLength = 0; /* Safety. */
+
+ if (pWav == NULL) {
+ return DRWAV_INVALID_ARGS;
+ }
+
+ *pLength = pWav->totalPCMFrameCount;
+
+ return DRWAV_SUCCESS;
+}
+
+
+DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData)
+{
+ size_t bytesWritten;
+
+ if (pWav == NULL || bytesToWrite == 0 || pData == NULL) {
+ return 0;
+ }
+
+ bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite);
+ pWav->dataChunkDataSize += bytesWritten;
+
+ return bytesWritten;
+}
+
+DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData)
+{
+ drwav_uint64 bytesToWrite;
+ drwav_uint64 bytesWritten;
+ const drwav_uint8* pRunningData;
+
+ if (pWav == NULL || framesToWrite == 0 || pData == NULL) {
+ return 0;
+ }
+
+ bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8);
+ if (bytesToWrite > DRWAV_SIZE_MAX) {
+ return 0;
+ }
+
+ bytesWritten = 0;
+ pRunningData = (const drwav_uint8*)pData;
+
+ while (bytesToWrite > 0) {
+ size_t bytesJustWritten;
+ drwav_uint64 bytesToWriteThisIteration;
+
+ bytesToWriteThisIteration = bytesToWrite;
+ DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */
+
+ bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData);
+ if (bytesJustWritten == 0) {
+ break;
+ }
+
+ bytesToWrite -= bytesJustWritten;
+ bytesWritten += bytesJustWritten;
+ pRunningData += bytesJustWritten;
+ }
+
+ return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels;
+}
+
+DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData)
+{
+ drwav_uint64 bytesToWrite;
+ drwav_uint64 bytesWritten;
+ drwav_uint32 bytesPerSample;
+ const drwav_uint8* pRunningData;
+
+ if (pWav == NULL || framesToWrite == 0 || pData == NULL) {
+ return 0;
+ }
+
+ bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8);
+ if (bytesToWrite > DRWAV_SIZE_MAX) {
+ return 0;
+ }
+
+ bytesWritten = 0;
+ pRunningData = (const drwav_uint8*)pData;
+
+ bytesPerSample = drwav_get_bytes_per_pcm_frame(pWav) / pWav->channels;
+ if (bytesPerSample == 0) {
+ return 0; /* Cannot determine bytes per sample, or bytes per sample is less than one byte. */
+ }
+
+ while (bytesToWrite > 0) {
+ drwav_uint8 temp[4096];
+ drwav_uint32 sampleCount;
+ size_t bytesJustWritten;
+ drwav_uint64 bytesToWriteThisIteration;
+
+ bytesToWriteThisIteration = bytesToWrite;
+ DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */
+
+ /*
+ WAV files are always little-endian. We need to byte swap on big-endian architectures. Since our input buffer is read-only we need
+ to use an intermediary buffer for the conversion.
+ */
+ sampleCount = sizeof(temp)/bytesPerSample;
+
+ if (bytesToWriteThisIteration > ((drwav_uint64)sampleCount)*bytesPerSample) {
+ bytesToWriteThisIteration = ((drwav_uint64)sampleCount)*bytesPerSample;
+ }
+
+ DRWAV_COPY_MEMORY(temp, pRunningData, (size_t)bytesToWriteThisIteration);
+ drwav__bswap_samples(temp, sampleCount, bytesPerSample);
+
+ bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, temp);
+ if (bytesJustWritten == 0) {
+ break;
+ }
+
+ bytesToWrite -= bytesJustWritten;
+ bytesWritten += bytesJustWritten;
+ pRunningData += bytesJustWritten;
+ }
+
+ return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels;
+}
+
+DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData)
+{
+ if (drwav__is_little_endian()) {
+ return drwav_write_pcm_frames_le(pWav, framesToWrite, pData);
+ } else {
+ return drwav_write_pcm_frames_be(pWav, framesToWrite, pData);
+ }
+}
+
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 totalFramesRead = 0;
+
+ static const drwav_int32 adaptationTable[] = {
+ 230, 230, 230, 230, 307, 409, 512, 614,
+ 768, 614, 512, 409, 307, 230, 230, 230
+ };
+ static const drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 };
+ static const drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 };
+
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(framesToRead > 0);
+
+ /* TODO: Lots of room for optimization here. */
+
+ while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
+ DRWAV_ASSERT(framesToRead > 0); /* This loop iteration will never get hit with framesToRead == 0 because it's asserted at the top, and we check for 0 inside the loop just below. */
+
+ /* If there are no cached frames we need to load a new block. */
+ if (pWav->msadpcm.cachedFrameCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) {
+ if (pWav->channels == 1) {
+ /* Mono. */
+ drwav_uint8 header[7];
+ if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
+ return totalFramesRead;
+ }
+ pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
+
+ pWav->msadpcm.predictor[0] = header[0];
+ pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 1);
+ pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 3);
+ pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 5);
+ pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][0];
+ pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[0][1];
+ pWav->msadpcm.cachedFrameCount = 2;
+
+ /* The predictor is used as an index into coeff1Table so we'll need to validate to ensure it never overflows. */
+ if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table)) {
+ return totalFramesRead; /* Invalid file. */
+ }
+ } else {
+ /* Stereo. */
+ drwav_uint8 header[14];
+ if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
+ return totalFramesRead;
+ }
+ pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
+
+ pWav->msadpcm.predictor[0] = header[0];
+ pWav->msadpcm.predictor[1] = header[1];
+ pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 2);
+ pWav->msadpcm.delta[1] = drwav_bytes_to_s16(header + 4);
+ pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 6);
+ pWav->msadpcm.prevFrames[1][1] = (drwav_int32)drwav_bytes_to_s16(header + 8);
+ pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 10);
+ pWav->msadpcm.prevFrames[1][0] = (drwav_int32)drwav_bytes_to_s16(header + 12);
+
+ pWav->msadpcm.cachedFrames[0] = pWav->msadpcm.prevFrames[0][0];
+ pWav->msadpcm.cachedFrames[1] = pWav->msadpcm.prevFrames[1][0];
+ pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][1];
+ pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[1][1];
+ pWav->msadpcm.cachedFrameCount = 2;
+
+ /* The predictor is used as an index into coeff1Table so we'll need to validate to ensure it never overflows. */
+ if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table) ||
+ pWav->msadpcm.predictor[1] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[1] >= drwav_countof(coeff2Table)) {
+ return totalFramesRead; /* Invalid file. */
+ }
+ }
+ }
+
+ /* Output anything that's cached. */
+ while (framesToRead > 0 && pWav->msadpcm.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
+ if (pBufferOut != NULL) {
+ drwav_uint32 iSample = 0;
+ for (iSample = 0; iSample < pWav->channels; iSample += 1) {
+ pBufferOut[iSample] = (drwav_int16)pWav->msadpcm.cachedFrames[(drwav_countof(pWav->msadpcm.cachedFrames) - (pWav->msadpcm.cachedFrameCount*pWav->channels)) + iSample];
+ }
+
+ pBufferOut += pWav->channels;
+ }
+
+ framesToRead -= 1;
+ totalFramesRead += 1;
+ pWav->readCursorInPCMFrames += 1;
+ pWav->msadpcm.cachedFrameCount -= 1;
+ }
+
+ if (framesToRead == 0) {
+ break;
+ }
+
+
+ /*
+ If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next
+ loop iteration which will trigger the loading of a new block.
+ */
+ if (pWav->msadpcm.cachedFrameCount == 0) {
+ if (pWav->msadpcm.bytesRemainingInBlock == 0) {
+ continue;
+ } else {
+ drwav_uint8 nibbles;
+ drwav_int32 nibble0;
+ drwav_int32 nibble1;
+
+ if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) {
+ return totalFramesRead;
+ }
+ pWav->msadpcm.bytesRemainingInBlock -= 1;
+
+ /* TODO: Optimize away these if statements. */
+ nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; }
+ nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; }
+
+ if (pWav->channels == 1) {
+ /* Mono. */
+ drwav_int32 newSample0;
+ drwav_int32 newSample1;
+
+ /* The predictor is read from the file and then indexed into a table. Check that it's in bounds. */
+ if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table)) {
+ return totalFramesRead;
+ }
+
+ newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
+ newSample0 += nibble0 * pWav->msadpcm.delta[0];
+ newSample0 = drwav_clamp(newSample0, -32768, 32767);
+
+ pWav->msadpcm.delta[0] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF);
+
+ pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
+ pWav->msadpcm.prevFrames[0][1] = newSample0;
+
+
+ newSample1 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
+ newSample1 += nibble1 * pWav->msadpcm.delta[0];
+ newSample1 = drwav_clamp(newSample1, -32768, 32767);
+
+ pWav->msadpcm.delta[0] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF);
+
+ pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
+ pWav->msadpcm.prevFrames[0][1] = newSample1;
+
+ pWav->msadpcm.cachedFrames[2] = newSample0;
+ pWav->msadpcm.cachedFrames[3] = newSample1;
+ pWav->msadpcm.cachedFrameCount = 2;
+ } else {
+ /* Stereo. */
+ drwav_int32 newSample0;
+ drwav_int32 newSample1;
+
+ /* Left. */
+ if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table)) {
+ return totalFramesRead; /* Out of bounds. Invalid file. */
+ }
+
+ newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
+ newSample0 += nibble0 * pWav->msadpcm.delta[0];
+ newSample0 = drwav_clamp(newSample0, -32768, 32767);
+
+ pWav->msadpcm.delta[0] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF);
+
+ pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
+ pWav->msadpcm.prevFrames[0][1] = newSample0;
+
+
+ /* Right. */
+ if (pWav->msadpcm.predictor[1] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[1] >= drwav_countof(coeff2Table)) {
+ return totalFramesRead; /* Out of bounds. Invalid file. */
+ }
+
+ newSample1 = ((pWav->msadpcm.prevFrames[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevFrames[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8;
+ newSample1 += nibble1 * pWav->msadpcm.delta[1];
+ newSample1 = drwav_clamp(newSample1, -32768, 32767);
+
+ pWav->msadpcm.delta[1] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8, 16, 0x7FFFFFFF);
+
+ pWav->msadpcm.prevFrames[1][0] = pWav->msadpcm.prevFrames[1][1];
+ pWav->msadpcm.prevFrames[1][1] = newSample1;
+
+ pWav->msadpcm.cachedFrames[2] = newSample0;
+ pWav->msadpcm.cachedFrames[3] = newSample1;
+ pWav->msadpcm.cachedFrameCount = 1;
+ }
+ }
+ }
+ }
+
+ return totalFramesRead;
+}
+
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 totalFramesRead = 0;
+ drwav_uint32 iChannel;
+
+ static const drwav_int32 indexTable[16] = {
+ -1, -1, -1, -1, 2, 4, 6, 8,
+ -1, -1, -1, -1, 2, 4, 6, 8
+ };
+
+ static const drwav_int32 stepTable[89] = {
+ 7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
+ 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
+ 50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
+ 130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
+ 337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
+ 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
+ 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
+ 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+ 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+ };
+
+ DRWAV_ASSERT(pWav != NULL);
+ DRWAV_ASSERT(framesToRead > 0);
+
+ /* TODO: Lots of room for optimization here. */
+
+ while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
+ DRWAV_ASSERT(framesToRead > 0); /* This loop iteration will never get hit with framesToRead == 0 because it's asserted at the top, and we check for 0 inside the loop just below. */
+
+ /* If there are no cached samples we need to load a new block. */
+ if (pWav->ima.cachedFrameCount == 0 && pWav->ima.bytesRemainingInBlock == 0) {
+ if (pWav->channels == 1) {
+ /* Mono. */
+ drwav_uint8 header[4];
+ if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
+ return totalFramesRead;
+ }
+ pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
+
+ if (header[2] >= drwav_countof(stepTable)) {
+ pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, DRWAV_SEEK_CUR);
+ pWav->ima.bytesRemainingInBlock = 0;
+ return totalFramesRead; /* Invalid data. */
+ }
+
+ pWav->ima.predictor[0] = (drwav_int16)drwav_bytes_to_u16(header + 0);
+ pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */
+ pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[0];
+ pWav->ima.cachedFrameCount = 1;
+ } else {
+ /* Stereo. */
+ drwav_uint8 header[8];
+ if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
+ return totalFramesRead;
+ }
+ pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
+
+ if (header[2] >= drwav_countof(stepTable) || header[6] >= drwav_countof(stepTable)) {
+ pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, DRWAV_SEEK_CUR);
+ pWav->ima.bytesRemainingInBlock = 0;
+ return totalFramesRead; /* Invalid data. */
+ }
+
+ pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0);
+ pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */
+ pWav->ima.predictor[1] = drwav_bytes_to_s16(header + 4);
+ pWav->ima.stepIndex[1] = drwav_clamp(header[6], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */
+
+ pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 2] = pWav->ima.predictor[0];
+ pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[1];
+ pWav->ima.cachedFrameCount = 1;
+ }
+ }
+
+ /* Output anything that's cached. */
+ while (framesToRead > 0 && pWav->ima.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
+ if (pBufferOut != NULL) {
+ drwav_uint32 iSample;
+ for (iSample = 0; iSample < pWav->channels; iSample += 1) {
+ pBufferOut[iSample] = (drwav_int16)pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + iSample];
+ }
+ pBufferOut += pWav->channels;
+ }
+
+ framesToRead -= 1;
+ totalFramesRead += 1;
+ pWav->readCursorInPCMFrames += 1;
+ pWav->ima.cachedFrameCount -= 1;
+ }
+
+ if (framesToRead == 0) {
+ break;
+ }
+
+ /*
+ If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next
+ loop iteration which will trigger the loading of a new block.
+ */
+ if (pWav->ima.cachedFrameCount == 0) {
+ if (pWav->ima.bytesRemainingInBlock == 0) {
+ continue;
+ } else {
+ /*
+ From what I can tell with stereo streams, it looks like every 4 bytes (8 samples) is for one channel. So it goes 4 bytes for the
+ left channel, 4 bytes for the right channel.
+ */
+ pWav->ima.cachedFrameCount = 8;
+ for (iChannel = 0; iChannel < pWav->channels; ++iChannel) {
+ drwav_uint32 iByte;
+ drwav_uint8 nibbles[4];
+ if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) {
+ pWav->ima.cachedFrameCount = 0;
+ return totalFramesRead;
+ }
+ pWav->ima.bytesRemainingInBlock -= 4;
+
+ for (iByte = 0; iByte < 4; ++iByte) {
+ drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0);
+ drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4);
+
+ drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]];
+ drwav_int32 predictor = pWav->ima.predictor[iChannel];
+
+ drwav_int32 diff = step >> 3;
+ if (nibble0 & 1) diff += step >> 2;
+ if (nibble0 & 2) diff += step >> 1;
+ if (nibble0 & 4) diff += step;
+ if (nibble0 & 8) diff = -diff;
+
+ predictor = drwav_clamp(predictor + diff, -32768, 32767);
+ pWav->ima.predictor[iChannel] = predictor;
+ pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1);
+ pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+0)*pWav->channels + iChannel] = predictor;
+
+
+ step = stepTable[pWav->ima.stepIndex[iChannel]];
+ predictor = pWav->ima.predictor[iChannel];
+
+ diff = step >> 3;
+ if (nibble1 & 1) diff += step >> 2;
+ if (nibble1 & 2) diff += step >> 1;
+ if (nibble1 & 4) diff += step;
+ if (nibble1 & 8) diff = -diff;
+
+ predictor = drwav_clamp(predictor + diff, -32768, 32767);
+ pWav->ima.predictor[iChannel] = predictor;
+ pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1);
+ pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+1)*pWav->channels + iChannel] = predictor;
+ }
+ }
+ }
+ }
+ }
+
+ return totalFramesRead;
+}
+
+
+#ifndef DR_WAV_NO_CONVERSION_API
+static const unsigned short g_drwavAlawTable[256] = {
+ 0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580,
+ 0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0,
+ 0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600,
+ 0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00,
+ 0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58,
+ 0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58,
+ 0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960,
+ 0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0,
+ 0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80,
+ 0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40,
+ 0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00,
+ 0x2B00, 0x2900, 0x2F00, 0x2D00, 0x2300, 0x2100, 0x2700, 0x2500, 0x3B00, 0x3900, 0x3F00, 0x3D00, 0x3300, 0x3100, 0x3700, 0x3500,
+ 0x0158, 0x0148, 0x0178, 0x0168, 0x0118, 0x0108, 0x0138, 0x0128, 0x01D8, 0x01C8, 0x01F8, 0x01E8, 0x0198, 0x0188, 0x01B8, 0x01A8,
+ 0x0058, 0x0048, 0x0078, 0x0068, 0x0018, 0x0008, 0x0038, 0x0028, 0x00D8, 0x00C8, 0x00F8, 0x00E8, 0x0098, 0x0088, 0x00B8, 0x00A8,
+ 0x0560, 0x0520, 0x05E0, 0x05A0, 0x0460, 0x0420, 0x04E0, 0x04A0, 0x0760, 0x0720, 0x07E0, 0x07A0, 0x0660, 0x0620, 0x06E0, 0x06A0,
+ 0x02B0, 0x0290, 0x02F0, 0x02D0, 0x0230, 0x0210, 0x0270, 0x0250, 0x03B0, 0x0390, 0x03F0, 0x03D0, 0x0330, 0x0310, 0x0370, 0x0350
+};
+
+static const unsigned short g_drwavMulawTable[256] = {
+ 0x8284, 0x8684, 0x8A84, 0x8E84, 0x9284, 0x9684, 0x9A84, 0x9E84, 0xA284, 0xA684, 0xAA84, 0xAE84, 0xB284, 0xB684, 0xBA84, 0xBE84,
+ 0xC184, 0xC384, 0xC584, 0xC784, 0xC984, 0xCB84, 0xCD84, 0xCF84, 0xD184, 0xD384, 0xD584, 0xD784, 0xD984, 0xDB84, 0xDD84, 0xDF84,
+ 0xE104, 0xE204, 0xE304, 0xE404, 0xE504, 0xE604, 0xE704, 0xE804, 0xE904, 0xEA04, 0xEB04, 0xEC04, 0xED04, 0xEE04, 0xEF04, 0xF004,
+ 0xF0C4, 0xF144, 0xF1C4, 0xF244, 0xF2C4, 0xF344, 0xF3C4, 0xF444, 0xF4C4, 0xF544, 0xF5C4, 0xF644, 0xF6C4, 0xF744, 0xF7C4, 0xF844,
+ 0xF8A4, 0xF8E4, 0xF924, 0xF964, 0xF9A4, 0xF9E4, 0xFA24, 0xFA64, 0xFAA4, 0xFAE4, 0xFB24, 0xFB64, 0xFBA4, 0xFBE4, 0xFC24, 0xFC64,
+ 0xFC94, 0xFCB4, 0xFCD4, 0xFCF4, 0xFD14, 0xFD34, 0xFD54, 0xFD74, 0xFD94, 0xFDB4, 0xFDD4, 0xFDF4, 0xFE14, 0xFE34, 0xFE54, 0xFE74,
+ 0xFE8C, 0xFE9C, 0xFEAC, 0xFEBC, 0xFECC, 0xFEDC, 0xFEEC, 0xFEFC, 0xFF0C, 0xFF1C, 0xFF2C, 0xFF3C, 0xFF4C, 0xFF5C, 0xFF6C, 0xFF7C,
+ 0xFF88, 0xFF90, 0xFF98, 0xFFA0, 0xFFA8, 0xFFB0, 0xFFB8, 0xFFC0, 0xFFC8, 0xFFD0, 0xFFD8, 0xFFE0, 0xFFE8, 0xFFF0, 0xFFF8, 0x0000,
+ 0x7D7C, 0x797C, 0x757C, 0x717C, 0x6D7C, 0x697C, 0x657C, 0x617C, 0x5D7C, 0x597C, 0x557C, 0x517C, 0x4D7C, 0x497C, 0x457C, 0x417C,
+ 0x3E7C, 0x3C7C, 0x3A7C, 0x387C, 0x367C, 0x347C, 0x327C, 0x307C, 0x2E7C, 0x2C7C, 0x2A7C, 0x287C, 0x267C, 0x247C, 0x227C, 0x207C,
+ 0x1EFC, 0x1DFC, 0x1CFC, 0x1BFC, 0x1AFC, 0x19FC, 0x18FC, 0x17FC, 0x16FC, 0x15FC, 0x14FC, 0x13FC, 0x12FC, 0x11FC, 0x10FC, 0x0FFC,
+ 0x0F3C, 0x0EBC, 0x0E3C, 0x0DBC, 0x0D3C, 0x0CBC, 0x0C3C, 0x0BBC, 0x0B3C, 0x0ABC, 0x0A3C, 0x09BC, 0x093C, 0x08BC, 0x083C, 0x07BC,
+ 0x075C, 0x071C, 0x06DC, 0x069C, 0x065C, 0x061C, 0x05DC, 0x059C, 0x055C, 0x051C, 0x04DC, 0x049C, 0x045C, 0x041C, 0x03DC, 0x039C,
+ 0x036C, 0x034C, 0x032C, 0x030C, 0x02EC, 0x02CC, 0x02AC, 0x028C, 0x026C, 0x024C, 0x022C, 0x020C, 0x01EC, 0x01CC, 0x01AC, 0x018C,
+ 0x0174, 0x0164, 0x0154, 0x0144, 0x0134, 0x0124, 0x0114, 0x0104, 0x00F4, 0x00E4, 0x00D4, 0x00C4, 0x00B4, 0x00A4, 0x0094, 0x0084,
+ 0x0078, 0x0070, 0x0068, 0x0060, 0x0058, 0x0050, 0x0048, 0x0040, 0x0038, 0x0030, 0x0028, 0x0020, 0x0018, 0x0010, 0x0008, 0x0000
+};
+
+static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn)
+{
+ return (short)g_drwavAlawTable[sampleIn];
+}
+
+static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn)
+{
+ return (short)g_drwavMulawTable[sampleIn];
+}
+
+
+
+DRWAV_PRIVATE void drwav__pcm_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
+{
+ size_t i;
+
+ /* Special case for 8-bit sample data because it's treated as unsigned. */
+ if (bytesPerSample == 1) {
+ drwav_u8_to_s16(pOut, pIn, totalSampleCount);
+ return;
+ }
+
+
+ /* Slightly more optimal implementation for common formats. */
+ if (bytesPerSample == 2) {
+ for (i = 0; i < totalSampleCount; ++i) {
+ *pOut++ = ((const drwav_int16*)pIn)[i];
+ }
+ return;
+ }
+ if (bytesPerSample == 3) {
+ drwav_s24_to_s16(pOut, pIn, totalSampleCount);
+ return;
+ }
+ if (bytesPerSample == 4) {
+ drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount);
+ return;
+ }
+
+
+ /* Anything more than 64 bits per sample is not supported. */
+ if (bytesPerSample > 8) {
+ DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
+ return;
+ }
+
+
+ /* Generic, slow converter. */
+ for (i = 0; i < totalSampleCount; ++i) {
+ drwav_uint64 sample = 0;
+ unsigned int shift = (8 - bytesPerSample) * 8;
+
+ unsigned int j;
+ for (j = 0; j < bytesPerSample; j += 1) {
+ DRWAV_ASSERT(j < 8);
+ sample |= (drwav_uint64)(pIn[j]) << shift;
+ shift += 8;
+ }
+
+ pIn += j;
+ *pOut++ = (drwav_int16)((drwav_int64)sample >> 48);
+ }
+}
+
+DRWAV_PRIVATE void drwav__ieee_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
+{
+ if (bytesPerSample == 4) {
+ drwav_f32_to_s16(pOut, (const float*)pIn, totalSampleCount);
+ return;
+ } else if (bytesPerSample == 8) {
+ drwav_f64_to_s16(pOut, (const double*)pIn, totalSampleCount);
+ return;
+ } else {
+ /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */
+ DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
+ return;
+ }
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ /* Fast path. */
+ if ((pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) || pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ if (pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, NULL);
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); /* Safe cast. */
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ if (pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, NULL);
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead);
+
+ /*
+ For some reason libsndfile seems to be returning samples of the opposite sign for a-law, but only
+ with AIFF files. For WAV files it seems to be the same as dr_wav. This is resulting in dr_wav's
+ automated tests failing. I'm not sure which is correct, but will assume dr_wav. If we're enforcing
+ libsndfile compatibility we'll swap the signs here.
+ */
+ #ifdef DR_WAV_LIBSNDFILE_COMPAT
+ {
+ if (pWav->container == drwav_container_aiff) {
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < samplesRead; iSample += 1) {
+ pBufferOut[iSample] = -pBufferOut[iSample];
+ }
+ }
+ }
+ #endif
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ if (pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, NULL);
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead);
+
+ /*
+ Just like with alaw, for some reason the signs between libsndfile and dr_wav are opposite. We just need to
+ swap the sign if we're compiling with libsndfile compatiblity so our automated tests don't fail.
+ */
+ #ifdef DR_WAV_LIBSNDFILE_COMPAT
+ {
+ if (pWav->container == drwav_container_aiff) {
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < samplesRead; iSample += 1) {
+ pBufferOut[iSample] = -pBufferOut[iSample];
+ }
+ }
+ }
+ #endif
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ if (pWav == NULL || framesToRead == 0) {
+ return 0;
+ }
+
+ if (pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, NULL);
+ }
+
+ /* Don't try to read more samples than can potentially fit in the output buffer. */
+ if (framesToRead * pWav->channels * sizeof(drwav_int16) > DRWAV_SIZE_MAX) {
+ framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int16) / pWav->channels;
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
+ return drwav_read_pcm_frames_s16__pcm(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
+ return drwav_read_pcm_frames_s16__ieee(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) {
+ return drwav_read_pcm_frames_s16__alaw(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
+ return drwav_read_pcm_frames_s16__mulaw(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
+ return drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ return drwav_read_pcm_frames_s16__ima(pWav, framesToRead, pBufferOut);
+ }
+
+ return 0;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut);
+ if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) {
+ drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels);
+ }
+
+ return framesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut);
+ if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) {
+ drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels);
+ }
+
+ return framesRead;
+}
+
+
+DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ int r;
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ int x = pIn[i];
+ r = x << 8;
+ r = r - 32768;
+ pOut[i] = (short)r;
+ }
+}
+
+DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ int r;
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ int x = ((int)(((unsigned int)(((const drwav_uint8*)pIn)[i*3+0]) << 8) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+1]) << 16) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+2])) << 24)) >> 8;
+ r = x >> 8;
+ pOut[i] = (short)r;
+ }
+}
+
+DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount)
+{
+ int r;
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ int x = pIn[i];
+ r = x >> 16;
+ pOut[i] = (short)r;
+ }
+}
+
+DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount)
+{
+ int r;
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ float x = pIn[i];
+ float c;
+ c = ((x < -1) ? -1 : ((x > 1) ? 1 : x));
+ c = c + 1;
+ r = (int)(c * 32767.5f);
+ r = r - 32768;
+ pOut[i] = (short)r;
+ }
+}
+
+DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount)
+{
+ int r;
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ double x = pIn[i];
+ double c;
+ c = ((x < -1) ? -1 : ((x > 1) ? 1 : x));
+ c = c + 1;
+ r = (int)(c * 32767.5);
+ r = r - 32768;
+ pOut[i] = (short)r;
+ }
+}
+
+DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ pOut[i] = drwav__alaw_to_s16(pIn[i]);
+ }
+}
+
+DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+ for (i = 0; i < sampleCount; ++i) {
+ pOut[i] = drwav__mulaw_to_s16(pIn[i]);
+ }
+}
+
+
+DRWAV_PRIVATE void drwav__pcm_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample)
+{
+ unsigned int i;
+
+ /* Special case for 8-bit sample data because it's treated as unsigned. */
+ if (bytesPerSample == 1) {
+ drwav_u8_to_f32(pOut, pIn, sampleCount);
+ return;
+ }
+
+ /* Slightly more optimal implementation for common formats. */
+ if (bytesPerSample == 2) {
+ drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount);
+ return;
+ }
+ if (bytesPerSample == 3) {
+ drwav_s24_to_f32(pOut, pIn, sampleCount);
+ return;
+ }
+ if (bytesPerSample == 4) {
+ drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount);
+ return;
+ }
+
+
+ /* Anything more than 64 bits per sample is not supported. */
+ if (bytesPerSample > 8) {
+ DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut));
+ return;
+ }
+
+
+ /* Generic, slow converter. */
+ for (i = 0; i < sampleCount; ++i) {
+ drwav_uint64 sample = 0;
+ unsigned int shift = (8 - bytesPerSample) * 8;
+
+ unsigned int j;
+ for (j = 0; j < bytesPerSample; j += 1) {
+ DRWAV_ASSERT(j < 8);
+ sample |= (drwav_uint64)(pIn[j]) << shift;
+ shift += 8;
+ }
+
+ pIn += j;
+ *pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0);
+ }
+}
+
+DRWAV_PRIVATE void drwav__ieee_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample)
+{
+ if (bytesPerSample == 4) {
+ unsigned int i;
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = ((const float*)pIn)[i];
+ }
+ return;
+ } else if (bytesPerSample == 8) {
+ drwav_f64_to_f32(pOut, (const double*)pIn, sampleCount);
+ return;
+ } else {
+ /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */
+ DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut));
+ return;
+ }
+}
+
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__pcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ /*
+ We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't
+ want to duplicate that code.
+ */
+ drwav_uint64 totalFramesRead;
+ drwav_int16 samples16[2048];
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels);
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */
+
+ pBufferOut += framesRead*pWav->channels;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ /* Fast path. */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) {
+ return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead);
+
+ #ifdef DR_WAV_LIBSNDFILE_COMPAT
+ {
+ if (pWav->container == drwav_container_aiff) {
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < samplesRead; iSample += 1) {
+ pBufferOut[iSample] = -pBufferOut[iSample];
+ }
+ }
+ }
+ #endif
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__mulaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead);
+
+ #ifdef DR_WAV_LIBSNDFILE_COMPAT
+ {
+ if (pWav->container == drwav_container_aiff) {
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < samplesRead; iSample += 1) {
+ pBufferOut[iSample] = -pBufferOut[iSample];
+ }
+ }
+ }
+ #endif
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ if (pWav == NULL || framesToRead == 0) {
+ return 0;
+ }
+
+ if (pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, NULL);
+ }
+
+ /* Don't try to read more samples than can potentially fit in the output buffer. */
+ if (framesToRead * pWav->channels * sizeof(float) > DRWAV_SIZE_MAX) {
+ framesToRead = DRWAV_SIZE_MAX / sizeof(float) / pWav->channels;
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
+ return drwav_read_pcm_frames_f32__pcm(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ return drwav_read_pcm_frames_f32__msadpcm_ima(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
+ return drwav_read_pcm_frames_f32__ieee(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) {
+ return drwav_read_pcm_frames_f32__alaw(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
+ return drwav_read_pcm_frames_f32__mulaw(pWav, framesToRead, pBufferOut);
+ }
+
+ return 0;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut);
+ if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) {
+ drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels);
+ }
+
+ return framesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut);
+ if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) {
+ drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels);
+ }
+
+ return framesRead;
+}
+
+
+DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+#ifdef DR_WAV_LIBSNDFILE_COMPAT
+ /*
+ It appears libsndfile uses slightly different logic for the u8 -> f32 conversion to dr_wav, which in my opinion is incorrect. It appears
+ libsndfile performs the conversion something like "f32 = (u8 / 256) * 2 - 1", however I think it should be "f32 = (u8 / 255) * 2 - 1" (note
+ the divisor of 256 vs 255). I use libsndfile as a benchmark for testing, so I'm therefore leaving this block here just for my automated
+ correctness testing. This is disabled by default.
+ */
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = (pIn[i] / 256.0f) * 2 - 1;
+ }
+#else
+ for (i = 0; i < sampleCount; ++i) {
+ float x = pIn[i];
+ x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */
+ x = x - 1; /* 0..2 to -1..1 */
+
+ *pOut++ = x;
+ }
+#endif
+}
+
+DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = pIn[i] * 0.000030517578125f;
+ }
+}
+
+DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ double x;
+ drwav_uint32 a = ((drwav_uint32)(pIn[i*3+0]) << 8);
+ drwav_uint32 b = ((drwav_uint32)(pIn[i*3+1]) << 16);
+ drwav_uint32 c = ((drwav_uint32)(pIn[i*3+2]) << 24);
+
+ x = (double)((drwav_int32)(a | b | c) >> 8);
+ *pOut++ = (float)(x * 0.00000011920928955078125);
+ }
+}
+
+DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount)
+{
+ size_t i;
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = (float)(pIn[i] / 2147483648.0);
+ }
+}
+
+DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = (float)pIn[i];
+ }
+}
+
+DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = drwav__alaw_to_s16(pIn[i]) / 32768.0f;
+ }
+}
+
+DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f;
+ }
+}
+
+
+
+DRWAV_PRIVATE void drwav__pcm_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
+{
+ unsigned int i;
+
+ /* Special case for 8-bit sample data because it's treated as unsigned. */
+ if (bytesPerSample == 1) {
+ drwav_u8_to_s32(pOut, pIn, totalSampleCount);
+ return;
+ }
+
+ /* Slightly more optimal implementation for common formats. */
+ if (bytesPerSample == 2) {
+ drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount);
+ return;
+ }
+ if (bytesPerSample == 3) {
+ drwav_s24_to_s32(pOut, pIn, totalSampleCount);
+ return;
+ }
+ if (bytesPerSample == 4) {
+ for (i = 0; i < totalSampleCount; ++i) {
+ *pOut++ = ((const drwav_int32*)pIn)[i];
+ }
+ return;
+ }
+
+
+ /* Anything more than 64 bits per sample is not supported. */
+ if (bytesPerSample > 8) {
+ DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
+ return;
+ }
+
+
+ /* Generic, slow converter. */
+ for (i = 0; i < totalSampleCount; ++i) {
+ drwav_uint64 sample = 0;
+ unsigned int shift = (8 - bytesPerSample) * 8;
+
+ unsigned int j;
+ for (j = 0; j < bytesPerSample; j += 1) {
+ DRWAV_ASSERT(j < 8);
+ sample |= (drwav_uint64)(pIn[j]) << shift;
+ shift += 8;
+ }
+
+ pIn += j;
+ *pOut++ = (drwav_int32)((drwav_int64)sample >> 32);
+ }
+}
+
+DRWAV_PRIVATE void drwav__ieee_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
+{
+ if (bytesPerSample == 4) {
+ drwav_f32_to_s32(pOut, (const float*)pIn, totalSampleCount);
+ return;
+ } else if (bytesPerSample == 8) {
+ drwav_f64_to_s32(pOut, (const double*)pIn, totalSampleCount);
+ return;
+ } else {
+ /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */
+ DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
+ return;
+ }
+}
+
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ /* Fast path. */
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) {
+ return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
+ }
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ /*
+ We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't
+ want to duplicate that code.
+ */
+ drwav_uint64 totalFramesRead = 0;
+ drwav_int16 samples16[2048];
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels);
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */
+
+ pBufferOut += framesRead*pWav->channels;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead);
+
+ #ifdef DR_WAV_LIBSNDFILE_COMPAT
+ {
+ if (pWav->container == drwav_container_aiff) {
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < samplesRead; iSample += 1) {
+ pBufferOut[iSample] = -pBufferOut[iSample];
+ }
+ }
+ }
+ #endif
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ drwav_uint64 totalFramesRead;
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0; /* Only byte-aligned formats are supported. */
+ }
+
+ totalFramesRead = 0;
+
+ while (framesToRead > 0) {
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
+ if (framesRead == 0) {
+ break;
+ }
+
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */
+
+ /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */
+ break;
+ }
+
+ drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead);
+
+ #ifdef DR_WAV_LIBSNDFILE_COMPAT
+ {
+ if (pWav->container == drwav_container_aiff) {
+ drwav_uint64 iSample;
+ for (iSample = 0; iSample < samplesRead; iSample += 1) {
+ pBufferOut[iSample] = -pBufferOut[iSample];
+ }
+ }
+ }
+ #endif
+
+ pBufferOut += samplesRead;
+ framesToRead -= framesRead;
+ totalFramesRead += framesRead;
+ }
+
+ return totalFramesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ if (pWav == NULL || framesToRead == 0) {
+ return 0;
+ }
+
+ if (pBufferOut == NULL) {
+ return drwav_read_pcm_frames(pWav, framesToRead, NULL);
+ }
+
+ /* Don't try to read more samples than can potentially fit in the output buffer. */
+ if (framesToRead * pWav->channels * sizeof(drwav_int32) > DRWAV_SIZE_MAX) {
+ framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int32) / pWav->channels;
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
+ return drwav_read_pcm_frames_s32__pcm(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ return drwav_read_pcm_frames_s32__msadpcm_ima(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
+ return drwav_read_pcm_frames_s32__ieee(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) {
+ return drwav_read_pcm_frames_s32__alaw(pWav, framesToRead, pBufferOut);
+ }
+
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
+ return drwav_read_pcm_frames_s32__mulaw(pWav, framesToRead, pBufferOut);
+ }
+
+ return 0;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut);
+ if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) {
+ drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels);
+ }
+
+ return framesRead;
+}
+
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+{
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut);
+ if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) {
+ drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels);
+ }
+
+ return framesRead;
+}
+
+
+DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = ((int)pIn[i] - 128) << 24;
+ }
+}
+
+DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = pIn[i] << 16;
+ }
+}
+
+DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ unsigned int s0 = pIn[i*3 + 0];
+ unsigned int s1 = pIn[i*3 + 1];
+ unsigned int s2 = pIn[i*3 + 2];
+
+ drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24));
+ *pOut++ = sample32;
+ }
+}
+
+DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = (drwav_int32)(2147483648.0f * pIn[i]);
+ }
+}
+
+DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]);
+ }
+}
+
+DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i = 0; i < sampleCount; ++i) {
+ *pOut++ = ((drwav_int32)drwav__alaw_to_s16(pIn[i])) << 16;
+ }
+}
+
+DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
+{
+ size_t i;
+
+ if (pOut == NULL || pIn == NULL) {
+ return;
+ }
+
+ for (i= 0; i < sampleCount; ++i) {
+ *pOut++ = ((drwav_int32)drwav__mulaw_to_s16(pIn[i])) << 16;
+ }
+}
+
+
+
+DRWAV_PRIVATE drwav_int16* drwav__read_pcm_frames_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount)
+{
+ drwav_uint64 sampleDataSize;
+ drwav_int16* pSampleData;
+ drwav_uint64 framesRead;
+
+ DRWAV_ASSERT(pWav != NULL);
+
+ /* Check for overflow before multiplication. */
+ if (pWav->channels == 0 || pWav->totalPCMFrameCount > DRWAV_SIZE_MAX / pWav->channels / sizeof(drwav_int16)) {
+ drwav_uninit(pWav);
+ return NULL; /* Overflow or invalid channels. */
+ }
+
+ sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int16);
+ if (sampleDataSize > DRWAV_SIZE_MAX) {
+ drwav_uninit(pWav);
+ return NULL; /* File's too big. */
+ }
+
+ pSampleData = (drwav_int16*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */
+ if (pSampleData == NULL) {
+ drwav_uninit(pWav);
+ return NULL; /* Failed to allocate memory. */
+ }
+
+ framesRead = drwav_read_pcm_frames_s16(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData);
+ if (framesRead != pWav->totalPCMFrameCount) {
+ drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks);
+ drwav_uninit(pWav);
+ return NULL; /* There was an error reading the samples. */
+ }
+
+ drwav_uninit(pWav);
+
+ if (sampleRate) {
+ *sampleRate = pWav->sampleRate;
+ }
+ if (channels) {
+ *channels = pWav->channels;
+ }
+ if (totalFrameCount) {
+ *totalFrameCount = pWav->totalPCMFrameCount;
+ }
+
+ return pSampleData;
+}
+
+DRWAV_PRIVATE float* drwav__read_pcm_frames_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount)
+{
+ drwav_uint64 sampleDataSize;
+ float* pSampleData;
+ drwav_uint64 framesRead;
+
+ DRWAV_ASSERT(pWav != NULL);
+
+ /* Check for overflow before multiplication. */
+ if (pWav->channels == 0 || pWav->totalPCMFrameCount > DRWAV_SIZE_MAX / pWav->channels / sizeof(float)) {
+ drwav_uninit(pWav);
+ return NULL; /* Overflow or invalid channels. */
+ }
+
+ sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(float);
+ if (sampleDataSize > DRWAV_SIZE_MAX) {
+ drwav_uninit(pWav);
+ return NULL; /* File's too big. */
+ }
+
+ pSampleData = (float*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */
+ if (pSampleData == NULL) {
+ drwav_uninit(pWav);
+ return NULL; /* Failed to allocate memory. */
+ }
+
+ framesRead = drwav_read_pcm_frames_f32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData);
+ if (framesRead != pWav->totalPCMFrameCount) {
+ drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks);
+ drwav_uninit(pWav);
+ return NULL; /* There was an error reading the samples. */
+ }
+
+ drwav_uninit(pWav);
+
+ if (sampleRate) {
+ *sampleRate = pWav->sampleRate;
+ }
+ if (channels) {
+ *channels = pWav->channels;
+ }
+ if (totalFrameCount) {
+ *totalFrameCount = pWav->totalPCMFrameCount;
+ }
+
+ return pSampleData;
+}
+
+DRWAV_PRIVATE drwav_int32* drwav__read_pcm_frames_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount)
+{
+ drwav_uint64 sampleDataSize;
+ drwav_int32* pSampleData;
+ drwav_uint64 framesRead;
+
+ DRWAV_ASSERT(pWav != NULL);
+
+ /* Check for overflow before multiplication. */
+ if (pWav->channels == 0 || pWav->totalPCMFrameCount > DRWAV_SIZE_MAX / pWav->channels / sizeof(drwav_int32)) {
+ drwav_uninit(pWav);
+ return NULL; /* Overflow or invalid channels. */
+ }
+
+ sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int32);
+ if (sampleDataSize > DRWAV_SIZE_MAX) {
+ drwav_uninit(pWav);
+ return NULL; /* File's too big. */
+ }
+
+ pSampleData = (drwav_int32*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */
+ if (pSampleData == NULL) {
+ drwav_uninit(pWav);
+ return NULL; /* Failed to allocate memory. */
+ }
+
+ framesRead = drwav_read_pcm_frames_s32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData);
+ if (framesRead != pWav->totalPCMFrameCount) {
+ drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks);
+ drwav_uninit(pWav);
+ return NULL; /* There was an error reading the samples. */
+ }
+
+ drwav_uninit(pWav);
+
+ if (sampleRate) {
+ *sampleRate = pWav->sampleRate;
+ }
+ if (channels) {
+ *channels = pWav->channels;
+ }
+ if (totalFrameCount) {
+ *totalFrameCount = pWav->totalPCMFrameCount;
+ }
+
+ return pSampleData;
+}
+
+
+
+DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init(&wav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init(&wav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init(&wav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+#ifndef DR_WAV_NO_STDIO
+DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+
+#ifndef DR_WAV_NO_WCHAR
+DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+#endif /* DR_WAV_NO_WCHAR */
+#endif /* DR_WAV_NO_STDIO */
+
+DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+
+DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav wav;
+
+ if (channelsOut) {
+ *channelsOut = 0;
+ }
+ if (sampleRateOut) {
+ *sampleRateOut = 0;
+ }
+ if (totalFrameCountOut) {
+ *totalFrameCountOut = 0;
+ }
+
+ if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) {
+ return NULL;
+ }
+
+ return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
+}
+#endif /* DR_WAV_NO_CONVERSION_API */
+
+
+DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pAllocationCallbacks != NULL) {
+ drwav__free_from_callbacks(p, pAllocationCallbacks);
+ } else {
+ drwav__free_default(p, NULL);
+ }
+}
+
+DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data)
+{
+ return ((drwav_uint16)data[0] << 0) | ((drwav_uint16)data[1] << 8);
+}
+
+DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data)
+{
+ return (drwav_int16)drwav_bytes_to_u16(data);
+}
+
+DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data)
+{
+ return drwav_bytes_to_u32_le(data);
+}
+
+DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data)
+{
+ union {
+ drwav_uint32 u32;
+ float f32;
+ } value;
+
+ value.u32 = drwav_bytes_to_u32(data);
+ return value.f32;
+}
+
+DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data)
+{
+ return (drwav_int32)drwav_bytes_to_u32(data);
+}
+
+DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data)
+{
+ return
+ ((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) |
+ ((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56);
+}
+
+DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data)
+{
+ return (drwav_int64)drwav_bytes_to_u64(data);
+}
+
+
+DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16])
+{
+ int i;
+ for (i = 0; i < 16; i += 1) {
+ if (a[i] != b[i]) {
+ return DRWAV_FALSE;
+ }
+ }
+
+ return DRWAV_TRUE;
+}
+
+DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b)
+{
+ return
+ a[0] == b[0] &&
+ a[1] == b[1] &&
+ a[2] == b[2] &&
+ a[3] == b[3];
+}
+
+#ifdef __MRC__
+/* Undo the pragma at the beginning of this file. */
+#pragma options opt reset
+#endif
+
+#endif /* dr_wav_c */
+#endif /* DR_WAV_IMPLEMENTATION */
diff --git a/cbb_RoomReverb/readme.md b/cbb_RoomReverb/readme.md
deleted file mode 100644
index 4b92d53..0000000
--- a/cbb_RoomReverb/readme.md
+++ /dev/null
@@ -1,2 +0,0 @@
-readme
-
diff --git a/cbb_RoomReverb/reverb.cpp b/cbb_RoomReverb/reverb.cpp
new file mode 100644
index 0000000..973494b
--- /dev/null
+++ b/cbb_RoomReverb/reverb.cpp
Binary files differ
diff --git a/cbb_RoomReverb/reverb.h b/cbb_RoomReverb/reverb.h
new file mode 100644
index 0000000..4e93c97
--- /dev/null
+++ b/cbb_RoomReverb/reverb.h
Binary files differ
diff --git a/cbb_RoomReverb/reverb_utils/AllpassDiffuser.h b/cbb_RoomReverb/reverb_utils/AllpassDiffuser.h
new file mode 100644
index 0000000..1646899
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/AllpassDiffuser.h
@@ -0,0 +1,149 @@
+
+#pragma once
+
+#include <vector>
+#include "ModulatedAllpass.h"
+#include "RandomBuffer.h"
+
+
+#ifndef BUFFER_SIZE
+#define BUFFER_SIZE 1024
+#endif
+
+
+
+namespace ReverbHallRoom
+{
+ class AllpassDiffuser
+ {
+ public:
+ static const int MaxStageCount = 12;
+
+ private:
+ int samplerate;
+
+ ModulatedAllpass filters[MaxStageCount];
+ int delay;
+ float modRate;
+ std::vector<float> seedValues;
+ int seed;
+ float crossSeed;
+
+ public:
+ int Stages;
+
+ AllpassDiffuser()
+ {
+ crossSeed = 0.0;
+ seed = 23456;
+ UpdateSeeds();
+ Stages = 1;
+
+ SetSamplerate(48000);
+ }
+
+ int GetSamplerate()
+ {
+ return samplerate;
+ }
+
+ void SetSamplerate(int samplerate)
+ {
+ this->samplerate = samplerate;
+ SetModRate(modRate);
+ }
+
+ void SetSeed(int seed)
+ {
+ this->seed = seed;
+ UpdateSeeds();
+ }
+
+ void SetCrossSeed(float crossSeed)
+ {
+ this->crossSeed = crossSeed;
+ UpdateSeeds();
+ }
+
+
+ bool GetModulationEnabled()
+ {
+ return filters[0].ModulationEnabled;
+ }
+
+ void SetModulationEnabled(bool value)
+ {
+ for (int i = 0; i < MaxStageCount; i++)
+ filters[i].ModulationEnabled = value;
+
+ }
+
+ void SetInterpolationEnabled(bool enabled)
+ {
+ for (int i = 0; i < MaxStageCount; i++)
+ filters[i].InterpolationEnabled = enabled;
+ }
+
+ void SetDelay(int delaySamples)
+ {
+ delay = delaySamples;
+ Update();
+ }
+
+ void SetFeedback(float feedback)
+ {
+ for (int i = 0; i < MaxStageCount; i++)
+ filters[i].Feedback = feedback;
+ }
+
+ void SetModAmount(float amount)
+ {
+ for (int i = 0; i < MaxStageCount; i++)
+ filters[i].ModAmount = amount * (0.85 + 0.3 * seedValues[MaxStageCount + i]);
+ }
+
+ void SetModRate(float rate)
+ {
+ modRate = rate;
+
+ for (int i = 0; i < MaxStageCount; i++)
+ filters[i].ModRate = rate * (0.85 + 0.3 * seedValues[MaxStageCount * 2 + i]) / samplerate;
+ }
+
+ void Process(float* input, float* output, int bufSize)
+ {
+ float tempBuffer[BUFFER_SIZE];
+
+ filters[0].Process(input, tempBuffer, bufSize);
+
+ for (int i = 1; i < Stages; i++)
+ filters[i].Process(tempBuffer, tempBuffer, bufSize);
+
+ Utils::Copy(output, tempBuffer, bufSize);
+ }
+
+ void ClearBuffers()
+ {
+ for (int i = 0; i < MaxStageCount; i++)
+ filters[i].ClearBuffers();
+ }
+
+ private:
+ void Update()
+ {
+ for (int i = 0; i < MaxStageCount; i++)
+ {
+ auto r = seedValues[i];
+ auto d = std::pow(10, r) * 0.1; // 0.1 ... 1.0
+ filters[i].SampleDelay = (int)(delay * d);
+ }
+ }
+
+ void UpdateSeeds()
+ {
+ this->seedValues = RandomBuffer::Generate(seed, MaxStageCount * 3, crossSeed);
+ Update();
+ }
+
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/Biquad.cpp b/cbb_RoomReverb/reverb_utils/Biquad.cpp
new file mode 100644
index 0000000..6f09c4a
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Biquad.cpp
@@ -0,0 +1,226 @@
+
+#include "Biquad.h"
+#define _USE_MATH_DEFINES
+#include "math.h"
+
+namespace ReverbHallRoom
+{
+ Biquad::Biquad()
+ {
+ ClearBuffers();
+ }
+
+ Biquad::Biquad(FilterType filterType, float fs)
+ {
+ Type = filterType;
+ SetSamplerate(fs);
+
+ SetGainDb(0.0f);
+ Frequency = (float)(fs * 0.25f);
+ SetQ(0.5f);
+ ClearBuffers();
+ }
+
+ Biquad::~Biquad()
+ {
+
+ }
+
+
+ float Biquad::GetSamplerate()
+ {
+ return fs;
+ }
+
+ void Biquad::SetSamplerate(float fs)
+ {
+ this->fs = fs;
+ fsInv = 1.0f / fs;
+ Update();
+ }
+
+ float Biquad::GetGainDb()
+ {
+ return gainDB;
+ }
+
+ float Biquad::GetGain()
+ {
+ return gain;
+ }
+
+ void Biquad::SetGainDb (float value)
+ {
+ // Clamp value between -60 and 60
+ if (value < -60)
+ value = -60;
+ else if (value > 60)
+ value = 60;
+
+ gainDB = value;
+ gain = powf (10.0f, value / 20.0f);
+ }
+
+ void Biquad::SetGain (float value)
+ {
+ if (value < 0.001f)
+ value = 0.001f; // -60dB
+ else if (value > 1000.0f)
+ value = 1000.0f; // 60dB
+
+ gain = value;
+ gainDB = log10f (gain) * 20;
+ }
+
+ float Biquad::GetQ()
+ {
+ return q;
+ }
+
+ void Biquad::SetQ(float value)
+ {
+ if (value < 0.001f)
+ value = 0.001f;
+ q = value;
+ }
+
+ // this is the newer set of formulas from http://www.earlevel.com/main/2011/01/02/biquad-formulas/
+ // Note that for shelf and peak filters, I had to invert the if/else statements for boost and cut, as
+ // I was getting the inverse desired effect, very odd...
+ void Biquad::Update()
+ {
+ auto Fc = Frequency;
+ //auto Fs = fs;
+
+ auto V = powf(10, fabsf(gainDB) / 20.0f);
+ //auto K = tanf(M_PI * Fc / Fs);
+ auto K = tanf(M_PI * Fc * fsInv);
+ auto Q = q;
+ double norm = 1.0;
+
+ switch (Type)
+ {
+ case FilterType::LowPass6db:
+ a1 = -expf(-2.0 * M_PI * (Fc * fsInv));
+ b0 = 1.0 + a1;
+ b1 = b2 = a2 = 0;
+ break;
+ case FilterType::HighPass6db:
+ a1 = -expf(-2.0 * M_PI * (Fc * fsInv));
+ b0 = a1;
+ b1 = -a1;
+ b2 = a2 = 0;
+ break;
+ case FilterType::LowPass:
+ norm = 1 / (1 + K / Q + K * K);
+ b0 = K * K * norm;
+ b1 = 2 * b0;
+ b2 = b0;
+ a1 = 2 * (K * K - 1) * norm;
+ a2 = (1 - K / Q + K * K) * norm;
+ break;
+ case FilterType::HighPass:
+ norm = 1 / (1 + K / Q + K * K);
+ b0 = 1 * norm;
+ b1 = -2 * b0;
+ b2 = b0;
+ a1 = 2 * (K * K - 1) * norm;
+ a2 = (1 - K / Q + K * K) * norm;
+ break;
+ case FilterType::BandPass:
+ norm = 1 / (1 + K / Q + K * K);
+ b0 = K / Q * norm;
+ b1 = 0;
+ b2 = -b0;
+ a1 = 2 * (K * K - 1) * norm;
+ a2 = (1 - K / Q + K * K) * norm;
+ break;
+ case FilterType::Notch:
+ norm = 1 / (1 + K / Q + K * K);
+ b0 = (1 + K * K) * norm;
+ b1 = 2 * (K * K - 1) * norm;
+ b2 = b0;
+ a1 = b1;
+ a2 = (1 - K / Q + K * K) * norm;
+ break;
+ case FilterType::Peak:
+ if (gainDB >= 0)
+ {
+ norm = 1 / (1 + 1 / Q * K + K * K);
+ b0 = (1 + V / Q * K + K * K) * norm;
+ b1 = 2 * (K * K - 1) * norm;
+ b2 = (1 - V / Q * K + K * K) * norm;
+ a1 = b1;
+ a2 = (1 - 1 / Q * K + K * K) * norm;
+ }
+ else
+ {
+ norm = 1 / (1 + V / Q * K + K * K);
+ b0 = (1 + 1 / Q * K + K * K) * norm;
+ b1 = 2 * (K * K - 1) * norm;
+ b2 = (1 - 1 / Q * K + K * K) * norm;
+ a1 = b1;
+ a2 = (1 - V / Q * K + K * K) * norm;
+ }
+ break;
+ case FilterType::LowShelf:
+ if (gainDB >= 0)
+ {
+ norm = 1 / (1 + sqrtf(2) * K + K * K);
+ b0 = (1 + sqrtf(2 * V) * K + V * K * K) * norm;
+ b1 = 2 * (V * K * K - 1) * norm;
+ b2 = (1 - sqrtf(2 * V) * K + V * K * K) * norm;
+ a1 = 2 * (K * K - 1) * norm;
+ a2 = (1 - sqrtf(2) * K + K * K) * norm;
+ }
+ else
+ {
+ norm = 1 / (1 + sqrtf(2 * V) * K + V * K * K);
+ b0 = (1 + sqrtf(2) * K + K * K) * norm;
+ b1 = 2 * (K * K - 1) * norm;
+ b2 = (1 - sqrtf(2) * K + K * K) * norm;
+ a1 = 2 * (V * K * K - 1) * norm;
+ a2 = (1 - sqrtf(2 * V) * K + V * K * K) * norm;
+ }
+ break;
+ case FilterType::HighShelf:
+ if (gainDB >= 0)
+ {
+ norm = 1 / (1 + sqrtf(2) * K + K * K);
+ b0 = (V + sqrtf(2 * V) * K + K * K) * norm;
+ b1 = 2 * (K * K - V) * norm;
+ b2 = (V - sqrtf(2 * V) * K + K * K) * norm;
+ a1 = 2 * (K * K - 1) * norm;
+ a2 = (1 - sqrtf(2) * K + K * K) * norm;
+ }
+ else
+ {
+ norm = 1 / (V + sqrtf(2 * V) * K + K * K);
+ b0 = (1 + sqrtf(2) * K + K * K) * norm;
+ b1 = 2 * (K * K - 1) * norm;
+ b2 = (1 - sqrtf(2) * K + K * K) * norm;
+ a1 = 2 * (K * K - V) * norm;
+ a2 = (V - sqrtf(2 * V) * K + K * K) * norm;
+ }
+ break;
+ }
+ }
+
+ double Biquad::GetResponse(float freq) const
+ {
+ double phi = powf((sinf(2 * M_PI * freq / (2.0 * fs))), 2);
+ double y = ((powf(b0 + b1 + b2, 2.0) - 4.0 * (b0 * b1 + 4.0 * b0 * b2 + b1 * b2) * phi + 16.0 * b0 * b2 * phi * phi) / (powf(1.0 + a1 + a2, 2.0) - 4.0 * (a1 + 4.0 * a2 + a1 * a2) * phi + 16.0 * a2 * phi * phi));
+ // y gives you power gain, not voltage gain, and this a 10 * log_10(g) formula instead of 20 * log_10(g)
+ // by taking the sqrt we get a value that's more suitable for signal processing, i.e. the voltage gain
+ return sqrtf(y);
+ }
+
+ void Biquad::ClearBuffers()
+ {
+ y = 0;
+ x2 = 0;
+ y2 = 0;
+ x1 = 0;
+ y1 = 0;
+ }
+}
diff --git a/cbb_RoomReverb/reverb_utils/Biquad.h b/cbb_RoomReverb/reverb_utils/Biquad.h
new file mode 100644
index 0000000..44d2496
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Biquad.h
@@ -0,0 +1,84 @@
+
+#pragma once
+
+namespace ReverbHallRoom
+{
+ class Biquad
+ {
+ public:
+ enum class FilterType
+ {
+ LowPass6db = 0,
+ HighPass6db,
+ LowPass,
+ HighPass,
+ BandPass,
+ Notch,
+ Peak,
+ LowShelf,
+ HighShelf
+ };
+
+ private:
+ float fs;
+ float fsInv;
+ float gainDB;
+ float q;
+ float a0, a1, a2, b0, b1, b2;
+ float x1, x2, y, y1, y2;
+ float gain;
+
+ public:
+ FilterType Type;
+ float Output;
+ float Frequency;
+
+ Biquad();
+ Biquad(FilterType filterType, float fs);
+ ~Biquad();
+
+ float GetSamplerate();
+ void SetSamplerate(float fs);
+ float GetGainDb();
+ void SetGainDb(float value);
+ float GetGain();
+ void SetGain(float value);
+ float GetQ();
+ void SetQ(float value);
+
+ void Update();
+
+ double GetResponse(float freq) const;
+
+ float inline Process(float x)
+ {
+ y = b0 * x + b1 * x1 + b2 * x2 - a1 * y1 - a2 * y2;
+ x2 = x1;
+ y2 = y1;
+ x1 = x;
+ y1 = y;
+
+ Output = y;
+ return Output;
+ }
+
+ void inline Process(float* input, float* output, int len)
+ {
+ for (int i = 0; i < len; i++)
+ {
+ float x = input[i];
+ y = ((b0 * x) + (b1 * x1) + (b2 * x2)) - (a1 * y1) - (a2 * y2);
+ x2 = x1;
+ y2 = y1;
+ x1 = x;
+ y1 = y;
+
+ output[i] = y;
+ }
+
+ Output = y;
+ }
+
+ void ClearBuffers();
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/DelayLine.h b/cbb_RoomReverb/reverb_utils/DelayLine.h
new file mode 100644
index 0000000..919161d
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/DelayLine.h
@@ -0,0 +1,263 @@
+
+#pragma once
+
+#include "Lp1.h"
+#include "ModulatedDelay.h"
+#include "AllpassDiffuser.h"
+#include "Biquad.h"
+
+namespace ReverbHallRoom
+{
+ template<unsigned int N>
+ class CircularBuffer
+ {
+ float buffer[N];
+ int idxRead;
+ int idxWrite;
+ int count;
+ public:
+ CircularBuffer()
+ {
+ Reset();
+ }
+
+ void Reset()
+ {
+ for (int i = 0; i < N; i++)
+ buffer[i] = 0.0f;
+ idxRead = 0;
+ idxWrite = 0;
+ count = 0;
+ }
+
+ int GetCount()
+ {
+ return count;
+ }
+
+ int PushZeros(float* data, int bufSize)
+ {
+ int countBefore = count;
+ for (int i = 0; i < bufSize; i++)
+ {
+ buffer[idxWrite] = 0.0f;
+ idxWrite = (idxWrite + 1) % N;
+ count++;
+ if (count >= N)
+ break; // overflow
+ }
+
+ return count - countBefore;
+ }
+
+ int Push(float* data, int bufSize)
+ {
+ int countBefore = count;
+ for (int i = 0; i < bufSize; i++)
+ {
+ buffer[idxWrite] = data[i];
+ idxWrite = (idxWrite + 1) % N;
+ count++;
+ if (count >= N)
+ break; // overflow
+ }
+
+ return count - countBefore;
+ }
+
+ int Pop(float* destination, int bufSize)
+ {
+ int countBefore = count;
+ for (int i = 0; i < bufSize; i++)
+ {
+ if (count > 0)
+ {
+ destination[i] = buffer[idxRead];
+ idxRead = (idxRead + 1) % N;
+ count--;
+ }
+ else
+ {
+ destination[i] = 0.0f;
+ }
+ }
+
+ return countBefore - count;
+ }
+
+ };
+
+ class DelayLine
+ {
+ private:
+ ModulatedDelay delay;
+ AllpassDiffuser diffuser;
+ Biquad lowShelf;
+ Biquad highShelf;
+ Lp1 lowPass;
+ CircularBuffer<2*BUFFER_SIZE> feedbackBuffer;
+ float feedback;
+
+ public:
+ bool DiffuserEnabled;
+ bool LowShelfEnabled;
+ bool HighShelfEnabled;
+ bool CutoffEnabled;
+ bool TapPostDiffuser;
+
+ DelayLine() :
+ lowShelf(Biquad::FilterType::LowShelf, 48000),
+ highShelf(Biquad::FilterType::HighShelf, 48000)
+ {
+ feedback = 0;
+
+ lowShelf.SetGainDb(-20);
+ lowShelf.Frequency = 20;
+
+ highShelf.SetGainDb(-20);
+ highShelf.Frequency = 19000;
+
+ lowPass.SetCutoffHz(1000);
+ lowShelf.Update();
+ highShelf.Update();
+ SetSamplerate(48000);
+ SetDiffuserSeed(1, 0.0);
+ }
+
+ void SetSamplerate(int samplerate)
+ {
+ diffuser.SetSamplerate(samplerate);
+ lowPass.SetSamplerate(samplerate);
+ lowShelf.SetSamplerate(samplerate);
+ highShelf.SetSamplerate(samplerate);
+ }
+
+ void SetDiffuserSeed(int seed, float crossSeed)
+ {
+ diffuser.SetSeed(seed);
+ diffuser.SetCrossSeed(crossSeed);
+ }
+
+ void SetDelay(int delaySamples)
+ {
+ delay.SampleDelay = delaySamples;
+ }
+
+ void SetFeedback(float feedb)
+ {
+ feedback = feedb;
+ }
+
+ void SetDiffuserDelay(int delaySamples)
+ {
+ diffuser.SetDelay(delaySamples);
+ }
+
+ void SetDiffuserFeedback(float feedb)
+ {
+ diffuser.SetFeedback(feedb);
+ }
+
+ void SetDiffuserStages(int stages)
+ {
+ diffuser.Stages = stages;
+ }
+
+ void SetLowShelfGain(float gainDb)
+ {
+ lowShelf.SetGainDb(gainDb);
+ lowShelf.Update();
+ }
+
+ void SetLowShelfFrequency(float frequency)
+ {
+ lowShelf.Frequency = frequency;
+ lowShelf.Update();
+ }
+
+ void SetHighShelfGain(float gainDb)
+ {
+ highShelf.SetGainDb(gainDb);
+ highShelf.Update();
+ }
+
+ void SetHighShelfFrequency(float frequency)
+ {
+ highShelf.Frequency = frequency;
+ highShelf.Update();
+ }
+
+ void SetCutoffFrequency(float frequency)
+ {
+ lowPass.SetCutoffHz(frequency);
+ }
+
+ void SetLineModAmount(float amount)
+ {
+ delay.ModAmount = amount;
+ }
+
+ void SetLineModRate(float rate)
+ {
+ delay.ModRate = rate;
+ }
+
+ void SetDiffuserModAmount(float amount)
+ {
+ diffuser.SetModulationEnabled(amount > 0.0);
+ diffuser.SetModAmount(amount);
+ }
+
+ void SetDiffuserModRate(float rate)
+ {
+ diffuser.SetModRate(rate);
+ }
+
+ void SetInterpolationEnabled(bool value)
+ {
+ diffuser.SetInterpolationEnabled(value);
+ }
+
+ void Process(float* input, float* output, int bufSize)
+ {
+ float tempBuffer[BUFFER_SIZE];
+ feedbackBuffer.Pop(tempBuffer, bufSize);
+
+ for (int i = 0; i < bufSize; i++)
+ tempBuffer[i] = input[i] + tempBuffer[i] * feedback;
+
+ delay.Process(tempBuffer, tempBuffer, bufSize);
+
+ if (!TapPostDiffuser)
+ Utils::Copy(output, tempBuffer, bufSize);
+ if (DiffuserEnabled)
+ diffuser.Process(tempBuffer, tempBuffer, bufSize);
+ if (LowShelfEnabled)
+ lowShelf.Process(tempBuffer, tempBuffer, bufSize);
+ if (HighShelfEnabled)
+ highShelf.Process(tempBuffer, tempBuffer, bufSize);
+ if (CutoffEnabled)
+ lowPass.Process(tempBuffer, tempBuffer, bufSize);
+
+ feedbackBuffer.Push(tempBuffer, bufSize);
+
+ if (TapPostDiffuser)
+ Utils::Copy(output, tempBuffer, bufSize);
+ }
+
+ void ClearDiffuserBuffer()
+ {
+ diffuser.ClearBuffers();
+ }
+
+ void ClearBuffers()
+ {
+ delay.ClearBuffers();
+ diffuser.ClearBuffers();
+ lowShelf.ClearBuffers();
+ highShelf.ClearBuffers();
+ lowPass.Output = 0;
+ feedbackBuffer.Reset();
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/Hp1.h b/cbb_RoomReverb/reverb_utils/Hp1.h
new file mode 100644
index 0000000..efb7b99
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Hp1.h
@@ -0,0 +1,91 @@
+
+#pragma once
+
+#define _USE_MATH_DEFINES
+#include <cmath>
+
+namespace ReverbHallRoom
+{
+ class Hp1
+ {
+ private:
+ float fs;
+ float b0, a1;
+ float lpOut;
+ float cutoffHz;
+
+ public:
+ float Output;
+
+ Hp1()
+ {
+ fs = 48000;
+ b0 = 1;
+ a1 = 0;
+ lpOut = 0.0;
+ cutoffHz = 100;
+ }
+
+ float GetSamplerate()
+ {
+ return fs;
+ }
+
+ void SetSamplerate(float samplerate)
+ {
+ fs = samplerate;
+ }
+
+ float GetCutoffHz()
+ {
+ return cutoffHz;
+ }
+
+ void SetCutoffHz(float hz)
+ {
+ cutoffHz = hz;
+ Update();
+ }
+
+ void ClearBuffers()
+ {
+ lpOut = 0;
+ Output = 0;
+ }
+
+ void Update()
+ {
+ // Prevent going over the Nyquist frequency
+ if (cutoffHz >= fs * 0.5f)
+ cutoffHz = fs * 0.499f;
+
+ auto x = 2.0f * M_PI * cutoffHz / fs;
+ auto nn = (2.0f - cosf(x));
+ auto alpha = nn - sqrtf(nn * nn - 1);
+
+ a1 = alpha;
+ b0 = 1 - alpha;
+ }
+
+ float Process(float input)
+ {
+ if (input == 0 && lpOut < 0.000001f)
+ {
+ Output = 0;
+ }
+ else
+ {
+ lpOut = b0 * input + a1 * lpOut;
+ Output = input - lpOut;
+ }
+
+ return Output;
+ }
+
+ void Process(float* input, float* output, int len)
+ {
+ for (int i = 0; i < len; i++)
+ output[i] = Process(input[i]);
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/LcgRandom.h b/cbb_RoomReverb/reverb_utils/LcgRandom.h
new file mode 100644
index 0000000..feffb19
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/LcgRandom.h
@@ -0,0 +1,76 @@
+
+#pragma once
+
+#include <stdint.h>
+
+namespace ReverbHallRoom
+{
+ class LcgRandom
+ {
+ private:
+ uint64_t x;
+ uint64_t a;
+ uint64_t c;
+
+ double doubleInv;
+ float floatUintInv;
+ float floatIntInv;
+
+ public:
+ inline LcgRandom(uint64_t seed = 0)
+ {
+ x = seed;
+ a = 22695477;
+ c = 1;
+
+ doubleInv = 1.0 / (double)UINT32_MAX;
+ floatUintInv = 1.0 / (float)UINT32_MAX;
+ floatIntInv = 1.0 / (float)INT32_MAX;
+ }
+
+ inline void SetSeed(uint64_t seed)
+ {
+ x = seed;
+ }
+
+ inline uint32_t NextUInt()
+ {
+ uint64_t axc = a * x + c;
+ //x = axc % m;
+ x = axc & 0xFFFFFFFF;
+ return (uint32_t)x;
+ }
+
+ inline int32_t NextInt()
+ {
+ int64_t axc = a * x + c;
+ //x = axc % m;
+ x = axc & 0x7FFFFFFF;
+ return (int32_t)x;
+ }
+
+ inline double NextDouble()
+ {
+ auto n = NextUInt();
+ return n * doubleInv;
+ }
+
+ inline float NextFloat()
+ {
+ auto n = NextInt();
+ return n * floatIntInv;
+ }
+
+ inline void GetFloats(float* buffer, int len)
+ {
+ for (int i = 0; i < len; i++)
+ buffer[i] = NextFloat();
+ }
+
+ inline void GetFloatsBipolar(float* buffer, int len)
+ {
+ for (int i = 0; i < len; i++)
+ buffer[i] = NextFloat() * 2 - 1;
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/Lp1.h b/cbb_RoomReverb/reverb_utils/Lp1.h
new file mode 100644
index 0000000..b90ee50
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Lp1.h
@@ -0,0 +1,86 @@
+
+#pragma once
+
+#define _USE_MATH_DEFINES
+#include <cmath>
+
+namespace ReverbHallRoom
+{
+ class Lp1
+ {
+ private:
+ float fs;
+ float b0, a1;
+ float cutoffHz;
+
+ public:
+ float Output;
+
+ Lp1()
+ {
+ fs = 48000;
+ b0 = 1;
+ a1 = 0;
+ cutoffHz = 1000;
+ }
+
+ float GetSamplerate()
+ {
+ return fs;
+ }
+
+ void SetSamplerate(float samplerate)
+ {
+ fs = samplerate;
+ }
+
+ float GetCutoffHz()
+ {
+ return cutoffHz;
+ }
+
+ void SetCutoffHz(float hz)
+ {
+ cutoffHz = hz;
+ Update();
+ }
+
+ void ClearBuffers()
+ {
+ Output = 0;
+ }
+
+ void Update()
+ {
+ // Prevent going over the Nyquist frequency
+ if (cutoffHz >= fs * 0.5f)
+ cutoffHz = fs * 0.499f;
+
+ auto x = 2.0f * M_PI * cutoffHz / fs;
+ auto nn = (2.0f - cosf(x));
+ auto alpha = nn - sqrtf(nn * nn - 1);
+
+ a1 = alpha;
+ b0 = 1 - alpha;
+ }
+
+ float Process(float input)
+ {
+ if (input == 0 && Output < 0.0000001f)
+ {
+ Output = 0;
+ }
+ else
+ {
+ Output = b0 * input + a1 * Output;
+ }
+ return Output;
+ }
+
+ void Process(float* input, float* output, int len)
+ {
+ for (int i = 0; i < len; i++)
+ output[i] = Process(input[i]);
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/ModulatedAllpass.h b/cbb_RoomReverb/reverb_utils/ModulatedAllpass.h
new file mode 100644
index 0000000..e08f4ad
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/ModulatedAllpass.h
@@ -0,0 +1,168 @@
+
+#pragma once
+
+#include "ModulatedAllpass.h"
+#include "Utils.h"
+#include <cmath>
+
+namespace ReverbHallRoom
+{
+ class ModulatedAllpass
+ {
+ public:
+ static const int DelayBufferSize = 19200; // 100ms at 192Khz
+ static const int ModulationUpdateRate = 8;
+
+ private:
+ //float delayBuffer[DelayBufferSize] = { 0 };
+ std::vector<float> delayBuffer;
+
+ int index;
+ uint64_t samplesProcessed;
+
+ float modPhase;
+ int delayA;
+ int delayB;
+ float gainA;
+ float gainB;
+
+ public:
+
+ int SampleDelay;
+ float Feedback;
+ float ModAmount;
+ float ModRate;
+
+ bool InterpolationEnabled;
+ bool ModulationEnabled;
+
+ ModulatedAllpass() : delayBuffer(DelayBufferSize, 0)
+ {
+ index = DelayBufferSize - 1;
+ samplesProcessed = 0;
+
+ modPhase = 0.01 + 0.98 * std::rand() / (float)RAND_MAX;
+ delayA = 0;
+ delayB = 0;
+ gainA = 0;
+ gainB = 0;
+
+ SampleDelay = 100;
+ Feedback = 0.5;
+ ModAmount = 0.0;
+ ModRate = 0.0;
+
+ InterpolationEnabled = true;
+ ModulationEnabled = true;
+ Update();
+ }
+
+ void ClearBuffers()
+ {
+ Utils::ZeroBuffer(delayBuffer.data(), DelayBufferSize);
+ }
+
+ void Process(float* input, float* output, int sampleCount)
+ {
+ if (ModulationEnabled)
+ ProcessWithMod(input, output, sampleCount);
+ else
+ ProcessNoMod(input, output, sampleCount);
+ }
+
+ private:
+ void ProcessNoMod(float* input, float* output, int sampleCount)
+ {
+ auto delayedIndex = index - SampleDelay;
+ if (delayedIndex < 0) delayedIndex += DelayBufferSize;
+
+ for (int i = 0; i < sampleCount; i++)
+ {
+ auto bufOut = delayBuffer[delayedIndex];
+ auto inVal = input[i] + bufOut * Feedback;
+
+ delayBuffer[index] = inVal;
+ output[i] = bufOut - inVal * Feedback;
+
+ index++;
+ delayedIndex++;
+ if (index >= DelayBufferSize) index -= DelayBufferSize;
+ if (delayedIndex >= DelayBufferSize) delayedIndex -= DelayBufferSize;
+ samplesProcessed++;
+ }
+ }
+
+ void ProcessWithMod(float* input, float* output, int sampleCount)
+ {
+ for (int i = 0; i < sampleCount; i++)
+ {
+ if (samplesProcessed >= ModulationUpdateRate)
+ {
+ Update();
+ samplesProcessed = 0;
+ }
+
+ float bufOut;
+
+ if (InterpolationEnabled)
+ {
+ int idxA = index - delayA;
+ int idxB = index - delayB;
+ idxA += DelayBufferSize * (idxA < 0); // modulo
+ idxB += DelayBufferSize * (idxB < 0); // modulo
+
+ bufOut = delayBuffer[idxA] * gainA + delayBuffer[idxB] * gainB;
+ }
+ else
+ {
+ int idxA = index - delayA;
+ idxA += DelayBufferSize * (idxA < 0); // modulo
+ bufOut = delayBuffer[idxA];
+ }
+
+ auto inVal = input[i] + bufOut * Feedback;
+ delayBuffer[index] = inVal;
+ output[i] = bufOut - inVal * Feedback;
+
+ index++;
+ if (index >= DelayBufferSize) index -= DelayBufferSize;
+ samplesProcessed++;
+ }
+ }
+
+ inline float Get(int delay)
+ {
+ int idx = index - delay;
+ if (idx < 0)
+ idx += DelayBufferSize;
+
+ return delayBuffer[idx];
+ }
+
+ void Update()
+ {
+ modPhase += ModRate * ModulationUpdateRate;
+ if (modPhase > 1)
+ modPhase = std::fmod(modPhase, 1.0);
+
+ auto mod = std::sinf(modPhase * 2 * M_PI);
+
+ if (ModAmount >= SampleDelay) // don't modulate to negative value
+ ModAmount = SampleDelay - 1;
+
+
+ auto totalDelay = SampleDelay + ModAmount * mod;
+
+ if (totalDelay <= 0) // should no longer be required
+ totalDelay = 1;
+
+ delayA = (int)totalDelay;
+ delayB = (int)totalDelay + 1;
+
+ auto partial = totalDelay - delayA;
+
+ gainA = 1 - partial;
+ gainB = partial;
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/ModulatedDelay.h b/cbb_RoomReverb/reverb_utils/ModulatedDelay.h
new file mode 100644
index 0000000..3adc1bb
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/ModulatedDelay.h
@@ -0,0 +1,107 @@
+
+#pragma once
+
+//#include "ModulatedDelay.h"
+#include "Utils.h"
+#include <stdint.h>
+#include <vector>
+
+namespace ReverbHallRoom
+{
+ class ModulatedDelay
+ {
+ private:
+
+ static const int ModulationUpdateRate = 8;
+ static const int DelayBufferSize = 192000 * 2;
+
+ //float delayBuffer[DelayBufferSize] = { 0 };
+ std::vector<float> delayBuffer;
+ int writeIndex;
+ int readIndexA;
+ int readIndexB;
+ uint64_t samplesProcessed;
+
+ float modPhase;
+ float gainA;
+ float gainB;
+
+ public:
+ int SampleDelay;
+
+ float ModAmount;
+ float ModRate;
+
+ ModulatedDelay() : delayBuffer(DelayBufferSize, 0)
+ {
+ writeIndex = 0;
+ readIndexA = 0;
+ readIndexB = 0;
+ samplesProcessed = 0;
+
+ modPhase = 0.01 + 0.98 * (std::rand() / (float)RAND_MAX);
+ gainA = 0;
+ gainB = 0;
+
+ SampleDelay = 100;
+ ModAmount = 0.0;
+ ModRate = 0.0;
+
+ Update();
+ }
+
+ void Process(float* input, float* output, int bufSize)
+ {
+ for (int i = 0; i < bufSize; i++)
+ {
+ if (samplesProcessed >= ModulationUpdateRate)
+ {
+ Update();
+ samplesProcessed = 0;
+ }
+
+ delayBuffer[writeIndex] = input[i];
+ output[i] = delayBuffer[readIndexA] * gainA + delayBuffer[readIndexB] * gainB;
+
+ writeIndex++;
+ readIndexA++;
+ readIndexB++;
+ if (writeIndex >= DelayBufferSize) writeIndex -= DelayBufferSize;
+ if (readIndexA >= DelayBufferSize) readIndexA -= DelayBufferSize;
+ if (readIndexB >= DelayBufferSize) readIndexB -= DelayBufferSize;
+ samplesProcessed++;
+ }
+ }
+
+ void ClearBuffers()
+ {
+ //Utils::ZeroBuffer(delayBuffer, DelayBufferSize);
+ Utils::ZeroBuffer(delayBuffer.data(), DelayBufferSize);
+ }
+
+
+ private:
+ void Update()
+ {
+ modPhase += ModRate * ModulationUpdateRate;
+ if (modPhase > 1)
+ modPhase = fmod(modPhase, 1.0);
+
+ auto mod = sinf(modPhase * 2 * M_PI);
+ auto totalDelay = SampleDelay + ModAmount * mod;
+
+ auto delayA = (int)totalDelay;
+ auto delayB = (int)totalDelay + 1;
+
+ auto partial = totalDelay - delayA;
+
+ gainA = 1 - partial;
+ gainB = partial;
+
+ readIndexA = writeIndex - delayA;
+ readIndexB = writeIndex - delayB;
+ if (readIndexA < 0) readIndexA += DelayBufferSize;
+ if (readIndexB < 0) readIndexB += DelayBufferSize;
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/MultitapDelay.h b/cbb_RoomReverb/reverb_utils/MultitapDelay.h
new file mode 100644
index 0000000..2474b75
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/MultitapDelay.h
@@ -0,0 +1,130 @@
+
+#pragma once
+
+#include <vector>
+#include <memory>
+#include <array>
+#include <cmath>
+#include "Utils.h"
+#include "RandomBuffer.h"
+
+namespace ReverbHallRoom
+{
+ class MultitapDelay
+ {
+ public:
+ static const int MaxTaps = 256;
+ static const int DelayBufferSize = 192000 * 2;
+
+ private:
+ //float delayBuffer[DelayBufferSize] = { 0 };
+ std::vector<float> delayBuffer;
+
+ float tapGains[MaxTaps] = { 0 };
+ float tapPosition[MaxTaps] = { 0 };
+
+ std::vector<float> seedValues;
+
+ int writeIdx;
+ int seed;
+ float crossSeed;
+ int count;
+ float lengthSamples;
+ float decay;
+
+ public:
+ MultitapDelay() : delayBuffer(DelayBufferSize, 0)
+ {
+ writeIdx = 0;
+ seed = 0;
+ crossSeed = 0.0;
+ count = 1;
+ lengthSamples = 1000;
+ decay = 1.0;
+
+ UpdateSeeds();
+ }
+
+ void SetSeed(int seed)
+ {
+ this->seed = seed;
+ UpdateSeeds();
+ }
+
+ void SetCrossSeed(float crossSeed)
+ {
+ this->crossSeed = crossSeed;
+ UpdateSeeds();
+ }
+
+ void SetTapCount(int tapCount)
+ {
+ if (tapCount < 1) tapCount = 1;
+ count = tapCount;
+ Update();
+ }
+
+ void SetTapLength(int tapLengthSamples)
+ {
+ if (tapLengthSamples < 10) tapLengthSamples = 10;
+ lengthSamples = tapLengthSamples;
+ Update();
+ }
+
+ void SetTapDecay(float tapDecay)
+ {
+ decay = tapDecay;
+ }
+
+ void Process(float* input, float* output, int bufSize)
+ {
+ float lengthScaler = lengthSamples / (float)count;
+ float totalGain = 3.0 / std::sqrtf(1 + count);
+ totalGain *= (1 + decay * 2);
+
+ for (int i = 0; i < bufSize; i++)
+ {
+ delayBuffer[writeIdx] = input[i];
+ output[i] = 0;
+
+ for (int j = 0; j < count; j++)
+ {
+ float offset = tapPosition[j] * lengthScaler;
+ float decayEffective = std::expf(-offset / lengthSamples * 3.3) * decay + (1-decay);
+ int readIdx = writeIdx - (int)offset;
+ if (readIdx < 0) readIdx += DelayBufferSize;
+
+ output[i] += delayBuffer[readIdx] * tapGains[j] * decayEffective * totalGain;
+ }
+
+ writeIdx = (writeIdx + 1) % DelayBufferSize;
+ }
+ }
+
+ void ClearBuffers()
+ {
+ Utils::ZeroBuffer(delayBuffer.data(), DelayBufferSize);
+ }
+
+
+ private:
+ void Update()
+ {
+ int s = 0;
+ auto rand = [&]() {return seedValues[s++]; };
+
+ for (int i = 0; i < MaxTaps; i++)
+ {
+ float phase = rand() < 0.5 ? 1 : -1;
+ tapGains[i] = Utils::DB2Gainf(-20 + rand() * 20) * phase;
+ tapPosition[i] = i + rand();
+ }
+ }
+
+ void UpdateSeeds()
+ {
+ this->seedValues = RandomBuffer::Generate(seed, MaxTaps * 3, crossSeed);
+ Update();
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/Parameters.cpp b/cbb_RoomReverb/reverb_utils/Parameters.cpp
new file mode 100644
index 0000000..0baeb10
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Parameters.cpp
@@ -0,0 +1,112 @@
+
+#include "Parameters.h"
+
+namespace ReverbHallRoom
+{
+ const char* ParameterLabel[Parameter::COUNT] = {
+ "Dry",
+ "Early",
+ "Late",
+
+ "Interpolation",
+ "High_Cut_ON",
+ "Low_Cut_ON",
+ "Input Mix",
+ "High Cut",
+ "Low Cut",
+ "Cross Seed",
+
+ "Multitap_Delay_ON",
+ "Count",
+ "Pre-delay",
+ "Decay",
+ "Length",
+
+ "Early_Diffusion_ON",
+ "Diffusion Stages",
+ "Delay",
+ "Feedback",
+ "Mod Amt",
+ "Mod Rate",
+
+ "Late_Mode",
+ "Late_Diffusion_ON",
+ "Line_Count",
+ "Line_Size",
+ "Line_Mod_Amt",
+ "Line_Decay",
+ "Line_Mod_Rate",
+ "Diffusion Stages",
+ "Diffusion_Delay",
+ "Diffusion_Feedback",
+ "Diffusion_Mod Amt",
+ "Diffusion_Mod Rate",
+
+ "EQ_Low_Shelf_ON",
+ "EQ_High_Shelf_ON",
+ "EQ_Lowpass_ON",
+ "Low_Freq",
+ "Low Gain",
+ "High Freq",
+ "High Gain",
+ "LP_Cutoff_Freq",
+
+ "Tap Seed",
+ "Diffusion Seed",
+ "Delay Seed",
+ "Late Diffusion Seed",
+ /*
+ "Interpolation",
+ "High Cut",
+ "Low Cut",
+ "Input Mix",
+ "High Cut",
+ "Low Cut",
+ "Dry",
+ "Early",
+ "Late",
+
+ "Multitap Delay",
+ "Count",
+ "Decay",
+ "Pre-delay",
+ "Length",
+
+ "Diffusion",
+ "Diffusion Stages",
+ "Delay",
+ "Mod Amt",
+ "Feedback",
+ "Mod Rate",
+
+ "Mode",
+ "Line Count",
+ "Diffusion",
+ "Diffusion Stages",
+ "Size",
+ "Mod Amt",
+ "Delay",
+ "Mod Amt",
+ "Decay",
+ "Mod Rate",
+ "Feedback",
+ "Mod Rate",
+
+ "Low Shelf",
+ "High Shelf",
+ "Lowpass",
+ "Low Freq",
+ "High Freq",
+ "Cutoff",
+ "Low Gain",
+ "High Gain",
+ "Cross Seed",
+
+ "Tap Seed",
+ "Diffusion Seed",
+ "Delay Seed",
+ "Late Diffusion Seed",
+ */
+ };
+
+}
diff --git a/cbb_RoomReverb/reverb_utils/Parameters.h b/cbb_RoomReverb/reverb_utils/Parameters.h
new file mode 100644
index 0000000..7b4a934
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Parameters.h
@@ -0,0 +1,326 @@
+
+#pragma once
+
+#include <math.h>
+#include <stdio.h>
+#include <string.h>
+#include "Utils.h"
+#include <stdio.h>
+
+
+
+#ifndef MAX_STR_SIZE
+#define MAX_STR_SIZE 64
+#endif
+
+
+namespace ReverbHallRoom
+{
+ namespace Parameter
+ {
+ const int DryOut = 0;
+ const int EarlyOut = 1;
+ const int LateOut = 2;
+
+ const int Interpolation = 3;
+ const int HighCutEnabled = 4;
+ const int LowCutEnabled = 5;
+ const int InputMix = 6;
+ const int HighCut = 7;
+ const int LowCut = 8;
+ const int EqCrossSeed = 9;
+
+ const int TapEnabled = 10;
+ const int TapCount = 11;
+ const int TapPredelay = 12;
+ const int TapDecay = 13;
+ const int TapLength = 14;
+
+ const int EarlyDiffuseEnabled = 15;
+ const int EarlyDiffuseCount = 16;
+ const int EarlyDiffuseDelay = 17;
+ const int EarlyDiffuseFeedback = 18;
+ const int EarlyDiffuseModAmount = 19;
+ const int EarlyDiffuseModRate = 20;
+
+ const int LateMode = 21;
+ const int LateDiffuseEnabled = 22;
+ const int LateLineCount = 23;
+ const int LateLineSize = 24;
+ const int LateLineModAmount = 25;
+ const int LateLineDecay = 26;
+ const int LateLineModRate = 27;
+ const int LateDiffuseCount = 28;
+ const int LateDiffuseDelay = 29;
+ const int LateDiffuseFeedback = 30;
+ const int LateDiffuseModAmount = 31;
+ const int LateDiffuseModRate = 32;
+
+ const int EqLowShelfEnabled = 33;
+ const int EqHighShelfEnabled = 34;
+ const int EqLowpassEnabled = 35;
+ const int EqLowFreq = 36;
+ const int EqLowGain = 37;
+ const int EqHighFreq = 38;
+ const int EqHighGain = 39;
+ const int EqCutoff = 40;
+
+ const int SeedTap = 41;
+ const int SeedDiffusion = 42;
+ const int SeedDelay = 43;
+ const int SeedPostDiffusion = 44;
+
+ const int COUNT = 45;
+ /*
+ const int Interpolation = 0;
+ const int LowCutEnabled = 1;
+ const int HighCutEnabled = 2;
+ const int InputMix = 3;
+ const int LowCut = 4;
+ const int HighCut = 5;
+ const int DryOut = 6;
+ const int EarlyOut = 7;
+ const int LateOut = 8;
+
+ const int TapEnabled = 9;
+ const int TapCount = 10;
+ const int TapDecay = 11;
+ const int TapPredelay = 12;
+ const int TapLength = 13;
+
+ const int EarlyDiffuseEnabled = 14;
+ const int EarlyDiffuseCount = 15;
+ const int EarlyDiffuseDelay = 16;
+ const int EarlyDiffuseModAmount = 17;
+ const int EarlyDiffuseFeedback = 18;
+ const int EarlyDiffuseModRate = 19;
+
+ const int LateMode = 20;
+ const int LateLineCount = 21;
+ const int LateDiffuseEnabled = 22;
+ const int LateDiffuseCount = 23;
+ const int LateLineSize = 24;
+ const int LateLineModAmount = 25;
+ const int LateDiffuseDelay = 26;
+ const int LateDiffuseModAmount = 27;
+ const int LateLineDecay = 28;
+ const int LateLineModRate = 29;
+ const int LateDiffuseFeedback = 30;
+ const int LateDiffuseModRate = 31;
+
+ const int EqLowShelfEnabled = 32;
+ const int EqHighShelfEnabled = 33;
+ const int EqLowpassEnabled = 34;
+ const int EqLowFreq = 35;
+ const int EqHighFreq = 36;
+ const int EqCutoff = 37;
+ const int EqLowGain = 38;
+ const int EqHighGain = 39;
+ const int EqCrossSeed = 40;
+
+ const int SeedTap = 41;
+ const int SeedDiffusion = 42;
+ const int SeedDelay = 43;
+ const int SeedPostDiffusion = 44;
+
+ const int COUNT = 45;
+ */
+ };
+
+ extern const char* ParameterLabel[Parameter::COUNT];
+
+ inline double ScaleParam(double val, int index)
+ {
+ switch (index)
+ {
+ case Parameter::Interpolation:
+ case Parameter::LowCutEnabled:
+ case Parameter::HighCutEnabled:
+ case Parameter::TapEnabled:
+ case Parameter::LateDiffuseEnabled:
+ case Parameter::EqLowShelfEnabled:
+ case Parameter::EqHighShelfEnabled:
+ case Parameter::EqLowpassEnabled:
+ case Parameter::EarlyDiffuseEnabled:
+ return val < 0.5 ? 0.0 : 1.0;
+
+ case Parameter::InputMix:
+ case Parameter::EarlyDiffuseFeedback:
+ case Parameter::TapDecay:
+ case Parameter::LateDiffuseFeedback:
+ case Parameter::EqCrossSeed:
+ return val;
+
+ case Parameter::SeedTap:
+ case Parameter::SeedDiffusion:
+ case Parameter::SeedDelay:
+ case Parameter::SeedPostDiffusion:
+ return (int)floor(val * 999.999);
+
+ case Parameter::LowCut:
+ return 20 + Utils::Resp4oct(val) * 980;
+ case Parameter::HighCut:
+ return 400 + Utils::Resp4oct(val) * 19600;
+
+ case Parameter::DryOut:
+ case Parameter::EarlyOut:
+ case Parameter::LateOut:
+ return -30 + val * 30;
+
+ case Parameter::TapCount:
+ return (int)(1 + val * 255);
+ case Parameter::TapPredelay:
+ return Utils::Resp1dec(val) * 500;
+ case Parameter::TapLength:
+ return 10 + val * 990;
+
+ case Parameter::EarlyDiffuseCount:
+ return (int)(1 + val * 11.999);
+ case Parameter::EarlyDiffuseDelay:
+ return 10 + val * 90;
+ case Parameter::EarlyDiffuseModAmount:
+ return val * 2.5;
+ case Parameter::EarlyDiffuseModRate:
+ return Utils::Resp2dec(val) * 5;
+
+ case Parameter::LateMode:
+ return val < 0.5 ? 0.0 : 1.0;
+ case Parameter::LateLineCount:
+ return (int)(1 + val * 11.999);
+ case Parameter::LateDiffuseCount:
+ return (int)(1 + val * 7.999);
+ case Parameter::LateLineSize:
+ return 20 + Utils::Resp2dec(val) * 980;
+ case Parameter::LateLineModAmount:
+ return val * 2.5;
+ case Parameter::LateDiffuseDelay:
+ return 10 + val * 90;
+ case Parameter::LateDiffuseModAmount:
+ return val * 2.5;
+ case Parameter::LateLineDecay:
+ return 0.05 + Utils::Resp3dec(val) * 59.95;
+ case Parameter::LateLineModRate:
+ return Utils::Resp2dec(val) * 5;
+ case Parameter::LateDiffuseModRate:
+ return Utils::Resp2dec(val) * 5;
+
+ case Parameter::EqLowFreq:
+ return 20 + Utils::Resp3oct(val) * 980;
+ case Parameter::EqHighFreq:
+ return 400 + Utils::Resp4oct(val) * 19600;
+ case Parameter::EqCutoff:
+ return 400 + Utils::Resp4oct(val) * 19600;
+ case Parameter::EqLowGain:
+ return -20 + val * 20;
+ case Parameter::EqHighGain:
+ return -20 + val * 20;
+ }
+ return 0;
+ }
+
+ inline void FormatParameter(double val, int maxLen, int paramId, char* buffer)
+ {
+ double s = ScaleParam(val, paramId);
+
+ switch (paramId)
+ {
+ case Parameter::Interpolation:
+ case Parameter::HighCutEnabled:
+ case Parameter::LowCutEnabled:
+ case Parameter::TapEnabled:
+ case Parameter::LateDiffuseEnabled:
+ case Parameter::EqLowShelfEnabled:
+ case Parameter::EqHighShelfEnabled:
+ case Parameter::EqLowpassEnabled:
+ case Parameter::EarlyDiffuseEnabled:
+ if (ScaleParam(val, paramId) == 1)
+ strcpy_s(buffer, MAX_STR_SIZE, "ENABLED");
+ else
+ strcpy_s(buffer, MAX_STR_SIZE, "DISABLED");
+ break;
+
+ case Parameter::InputMix:
+ case Parameter::EarlyDiffuseFeedback:
+ case Parameter::TapDecay:
+ case Parameter::LateDiffuseFeedback:
+ case Parameter::EqCrossSeed:
+ snprintf(buffer, MAX_STR_SIZE, "%d%%", (int)(s * 100));
+ break;
+
+ case Parameter::SeedTap:
+ case Parameter::SeedDiffusion:
+ case Parameter::SeedDelay:
+ case Parameter::SeedPostDiffusion:
+ snprintf(buffer, MAX_STR_SIZE, "%03d", (int)s);
+ break;
+
+ case Parameter::LowCut:
+ case Parameter::HighCut:
+ case Parameter::EqLowFreq:
+ case Parameter::EqHighFreq:
+ case Parameter::EqCutoff:
+ snprintf(buffer, MAX_STR_SIZE, "%d Hz", (int)s);
+ break;
+
+ case Parameter::DryOut:
+ case Parameter::EarlyOut:
+ case Parameter::LateOut:
+ if (s <= -30)
+ strcpy_s(buffer, MAX_STR_SIZE, "MUTED");
+ else
+ snprintf(buffer, MAX_STR_SIZE, "%.1f dB", s);
+ break;
+
+ case Parameter::TapCount:
+ case Parameter::EarlyDiffuseCount:
+ case Parameter::LateLineCount:
+ case Parameter::LateDiffuseCount:
+ snprintf(buffer, MAX_STR_SIZE, "%d", (int)s);
+ break;
+
+ case Parameter::TapPredelay:
+ case Parameter::TapLength:
+ case Parameter::EarlyDiffuseDelay:
+ case Parameter::LateLineSize:
+ case Parameter::LateDiffuseDelay:
+ snprintf(buffer, MAX_STR_SIZE, "%d ms", (int)s);
+ break;
+
+ case Parameter::LateLineDecay:
+ if (s < 1)
+ snprintf(buffer, MAX_STR_SIZE, "%d ms", (int)(s * 1000));
+ else if (s < 10)
+ snprintf(buffer, MAX_STR_SIZE, "%.2f sec", s);
+ else
+ snprintf(buffer, MAX_STR_SIZE, "%.1f sec", s);
+ break;
+
+ case Parameter::LateMode:
+ if (s == 1)
+ strcpy_s(buffer, MAX_STR_SIZE, "POST");
+ else
+ strcpy_s(buffer, MAX_STR_SIZE, "PRE");
+ break;
+
+ case Parameter::EarlyDiffuseModAmount:
+ case Parameter::LateLineModAmount:
+ case Parameter::LateDiffuseModAmount:
+ snprintf(buffer, MAX_STR_SIZE, "%d%%", (int)(s * 100));
+ break;
+
+ case Parameter::EarlyDiffuseModRate:
+ case Parameter::LateLineModRate:
+ case Parameter::LateDiffuseModRate:
+ snprintf(buffer, MAX_STR_SIZE, "%.2f Hz", s);
+ break;
+
+ case Parameter::EqLowGain:
+ case Parameter::EqHighGain:
+ snprintf(buffer, MAX_STR_SIZE, "%.1f dB", s);
+ break;
+
+ default:
+ snprintf(buffer, MAX_STR_SIZE, "%.2f", s);
+ }
+ }
+}
diff --git a/cbb_RoomReverb/reverb_utils/Programs.h b/cbb_RoomReverb/reverb_utils/Programs.h
new file mode 100644
index 0000000..c2a5f14
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Programs.h
@@ -0,0 +1,116 @@
+
+#pragma once
+
+#include "Parameters.h"
+
+namespace ReverbHallRoom
+{
+ float ProgramDarkPlate[Parameter::COUNT];
+#if 1
+ void initPrograms(Reverb_common *common, Reverb_taps *taps, Reverb_early *early, Reverb_late *late, Reverb_eq *eq)
+ {
+ ProgramDarkPlate[Parameter::DryOut] = common->dry; //0.6f; // 骞插０姣斾緥閫備腑
+ ProgramDarkPlate[Parameter::EarlyOut] = common->early; //0.7f; // 鏃╂湡鍙嶅皠杈撳嚭澧炵泭
+ ProgramDarkPlate[Parameter::LateOut] = common->late; //0.8f; // 鍚庢湡娣峰搷杈撳嚭澧炵泭
+ ProgramDarkPlate[Parameter::Interpolation] = common->input_mix_on; //1.0f; // 鍚敤鎻掑�
+ ProgramDarkPlate[Parameter::HighCutEnabled] = common->high_cut_on; //1.0f; // 鍚敤楂樺垏
+ ProgramDarkPlate[Parameter::LowCutEnabled] = common->low_cut_on; //1.0f; // 鍚敤浣庡垏
+ ProgramDarkPlate[Parameter::InputMix] = common->input_mix; //0.25f; // 杈撳叆浜ゅ弶 25%
+ ProgramDarkPlate[Parameter::HighCut] = common->high_cut; //0.8f; // 楂樺垏绾�16kHz
+ ProgramDarkPlate[Parameter::LowCut] = common->low_cut; //0.4f; // 浣庡垏绾�400Hz
+ ProgramDarkPlate[Parameter::EqCrossSeed] = common->cross_seed; //0.5f; // 宸﹀彸鍧囪 涓瓑鐩稿叧鎬�+
+ ProgramDarkPlate[Parameter::TapEnabled] = taps->multi_taps_on; //1.0f; // 鍚敤鏃╂湡鍙嶅皠
+ ProgramDarkPlate[Parameter::TapCount] = taps->taps_count; //0.7f; // 鎶藉ご鏁扮害 180
+ ProgramDarkPlate[Parameter::TapPredelay] = taps->taps_delay; //0.4f; // 棰勫欢杩�200ms
+ ProgramDarkPlate[Parameter::TapDecay] = taps->taps_decay; //0.7f; // 鏃╂湡鍙嶅皠琛板噺
+ ProgramDarkPlate[Parameter::TapLength] = taps->taps_length; //0.8f; // 鏃╂湡鍙嶅皠闀垮害 800ms
+
+ ProgramDarkPlate[Parameter::EarlyDiffuseEnabled] = early->early_difus_on; //1.0f; // 鍚敤鏃╂湡鎵╂暎鍣�
+ ProgramDarkPlate[Parameter::EarlyDiffuseCount] = early->early_count; //0.4f; // 鏃╂湡鎵╂暎绾ф暟 5 绾у乏鍙�+ ProgramDarkPlate[Parameter::EarlyDiffuseDelay] = early->early_delay; //0.3f; // 绾�37ms
+ ProgramDarkPlate[Parameter::EarlyDiffuseFeedback] = early->early_feedback; //0.6f; // 鍙嶉 0.6
+ ProgramDarkPlate[Parameter::EarlyDiffuseModAmount] = early->early_mod_amt; //0.15f; // 杞诲井璋冨埗
+ ProgramDarkPlate[Parameter::EarlyDiffuseModRate] = early->early_mod_rate; //0.2f; // 绾�1Hz
+
+ ProgramDarkPlate[Parameter::LateMode] = late->late_mode; //1.0f; // POST 妯″紡锛屽彇鍐呴儴鎵╂暎鍚�+// ProgramDarkPlate[Parameter::LateDiffuseEnabled] = 0.5f;//late->late_reflect_on; //1.0f; // 鍚敤鍚庢湡鎵╂暎
+ ProgramDarkPlate[Parameter::LateDiffuseEnabled] = late->late_reflect_on; //1.0f; // 鍚敤鍚庢湡鎵╂暎
+ ProgramDarkPlate[Parameter::LateLineCount] = late->line_count; //0.9f; // 11 鏉″欢杩熺嚎
+ ProgramDarkPlate[Parameter::LateLineSize] = late->line_size; //0.7f; // 寤惰繜绾块暱搴︾害 700ms
+ ProgramDarkPlate[Parameter::LateLineModAmount] = late->line_mod_amt; //0.3f; // 璋冨埗娣卞害绾�0.75ms
+ ProgramDarkPlate[Parameter::LateLineDecay] = late->line_decay; //0.25f; // 娣峰搷鏃堕棿绾�2 绉掞紙鍚璋冩暣锛�+ ProgramDarkPlate[Parameter::LateLineModRate] = late->line_mod_rate; //0.25f; // 璋冨埗閫熺巼绾�1.25Hz
+ ProgramDarkPlate[Parameter::LateDiffuseCount] = late->difus_count; //0.6f; // 鍚庢湡鎵╂暎绾ф暟 5 绾�+// ProgramDarkPlate[Parameter::LateDiffuseDelay] = late->difus_delay; //0.3f; // 绾�37ms
+ ProgramDarkPlate[Parameter::LateDiffuseDelay] = late->difus_delay;
+ ProgramDarkPlate[Parameter::LateDiffuseFeedback] = late->difus_feedback; //0.7f; // 鍙嶉 0.7
+ ProgramDarkPlate[Parameter::LateDiffuseModAmount] = late->difus_mod_amt; //0.15f; // 杞诲井璋冨埗
+ ProgramDarkPlate[Parameter::LateDiffuseModRate] = late->difus_mod_rate; //0.2f; // 绾�1Hz
+
+ ProgramDarkPlate[Parameter::EqLowShelfEnabled] = eq->low_shelf_on; //0.0f; // 绂佺敤浣庢灦
+ ProgramDarkPlate[Parameter::EqHighShelfEnabled] = eq->high_shelf_on; //1.0f; // 鍚敤楂樻灦
+ ProgramDarkPlate[Parameter::EqLowpassEnabled] = eq->low_pass_on; //1.0f; // 鍚敤浣庨�
+ ProgramDarkPlate[Parameter::EqLowFreq] = eq->low_shelf_freq; //0.4f; // 浣庢灦杞姌 400Hz
+ ProgramDarkPlate[Parameter::EqLowGain] = eq->low_shelf_gain; //0.5f; // 浣庢灦澧炵泭 0dB 闄勮繎
+ ProgramDarkPlate[Parameter::EqHighFreq] = eq->high_shelf_freq; //0.6f; // 楂樻灦杞姌 10kHz
+ ProgramDarkPlate[Parameter::EqHighGain] = eq->high_shelf_gain; //0.7f; // 楂樻灦澧炵泭 +8dB
+ ProgramDarkPlate[Parameter::EqCutoff] = eq->low_pass_freq; //0.7f; // 浣庨�鎴绾�14kHz
+
+ ProgramDarkPlate[Parameter::SeedDelay] = 0.5f; // 闅忔満绉嶅瓙
+ ProgramDarkPlate[Parameter::SeedDiffusion] = 0.5f;
+ ProgramDarkPlate[Parameter::SeedPostDiffusion] = 0.5f;
+ ProgramDarkPlate[Parameter::SeedTap] = 0.5f;
+
+ }
+#else
+ void initPrograms()
+ {
+ ProgramDarkPlate[Parameter::DryOut] = 0.8705999851226807;
+ ProgramDarkPlate[Parameter::EarlyDiffuseCount] = 0.2960000038146973;
+ ProgramDarkPlate[Parameter::EarlyDiffuseDelay] = 0.3066999912261963;
+ ProgramDarkPlate[Parameter::EarlyDiffuseEnabled] = 0.0;
+ ProgramDarkPlate[Parameter::EarlyDiffuseFeedback] = 0.7706999778747559;
+ ProgramDarkPlate[Parameter::EarlyDiffuseModAmount] = 0.143899992108345;
+ ProgramDarkPlate[Parameter::EarlyDiffuseModRate] = 0.2466999888420105;
+ ProgramDarkPlate[Parameter::EarlyOut] = 0.3; //0.0
+ ProgramDarkPlate[Parameter::EqCrossSeed] = 0.0;
+ ProgramDarkPlate[Parameter::EqCutoff] = 0.9759999513626099;
+ ProgramDarkPlate[Parameter::EqHighFreq] = 0.5133999586105347;
+ ProgramDarkPlate[Parameter::EqHighGain] = 0.7680000066757202;
+ ProgramDarkPlate[Parameter::EqHighShelfEnabled] = 1.0;
+ ProgramDarkPlate[Parameter::EqLowFreq] = 0.3879999816417694;
+ ProgramDarkPlate[Parameter::EqLowGain] = 0.5559999942779541;
+ ProgramDarkPlate[Parameter::EqLowShelfEnabled] = 0.0;
+ ProgramDarkPlate[Parameter::EqLowpassEnabled] = 0.0;
+ ProgramDarkPlate[Parameter::HighCut] = 0.2933000028133392;
+ ProgramDarkPlate[Parameter::HighCutEnabled] = 0.0;
+ ProgramDarkPlate[Parameter::InputMix] = 0.2346999943256378;
+ ProgramDarkPlate[Parameter::Interpolation] = 1.0;
+ ProgramDarkPlate[Parameter::LateDiffuseCount] = 0.4879999756813049;
+ ProgramDarkPlate[Parameter::LateDiffuseDelay] = 0.239999994635582;
+ ProgramDarkPlate[Parameter::LateDiffuseEnabled] = 1.0;
+ ProgramDarkPlate[Parameter::LateDiffuseFeedback] = 0.8506999611854553;
+ ProgramDarkPlate[Parameter::LateDiffuseModAmount] = 0.1467999964952469;
+ ProgramDarkPlate[Parameter::LateDiffuseModRate] = 0.1666999906301498;
+ ProgramDarkPlate[Parameter::LateLineCount] = 1.0;
+ ProgramDarkPlate[Parameter::LateLineDecay] = 0.6345999836921692;
+ ProgramDarkPlate[Parameter::LateLineModAmount] = 0.2719999849796295;
+ ProgramDarkPlate[Parameter::LateLineModRate] = 0.2292999923229218;
+ ProgramDarkPlate[Parameter::LateLineSize] = 0.4693999886512756;
+ ProgramDarkPlate[Parameter::LateMode] = 1.0;
+ ProgramDarkPlate[Parameter::LateOut] = 0.6613999605178833;
+ ProgramDarkPlate[Parameter::LowCut] = 0.6399999856948853;
+ ProgramDarkPlate[Parameter::LowCutEnabled] = 1.0;
+ ProgramDarkPlate[Parameter::SeedDelay] = 0.2180999964475632;
+ ProgramDarkPlate[Parameter::SeedDiffusion] = 0.1850000023841858;
+ ProgramDarkPlate[Parameter::SeedPostDiffusion] = 0.3652999997138977;
+ ProgramDarkPlate[Parameter::SeedTap] = 0.3339999914169312;
+ ProgramDarkPlate[Parameter::TapDecay] = 1.0;
+ ProgramDarkPlate[Parameter::TapLength] = 0.9866999983787537;
+ ProgramDarkPlate[Parameter::TapPredelay] = 0.0;
+ ProgramDarkPlate[Parameter::TapCount] = 0.1959999948740005;
+ ProgramDarkPlate[Parameter::TapEnabled] = 0.0;
+ }
+#endif
+}
diff --git a/cbb_RoomReverb/reverb_utils/RandomBuffer.cpp b/cbb_RoomReverb/reverb_utils/RandomBuffer.cpp
new file mode 100644
index 0000000..7ebede0
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/RandomBuffer.cpp
@@ -0,0 +1,36 @@
+
+#include <climits>
+#include "RandomBuffer.h"
+#include "LcgRandom.h"
+
+namespace ReverbHallRoom
+{
+ std::vector<float> RandomBuffer::Generate(uint64_t seed, int count)
+ {
+ LcgRandom rand(seed);
+ std::vector<float> output;
+
+ for (int i = 0; i < count; i++)
+ {
+ unsigned int val = rand.NextUInt();
+ float fVal = val / (float)UINT_MAX;
+ output.push_back(fVal);
+ }
+
+ return output;
+ }
+
+ std::vector<float> RandomBuffer::Generate(uint64_t seed, int count, float crossSeed)
+ {
+ auto seedA = seed;
+ auto seedB = ~seed;
+ auto seriesA = Generate(seedA, count);
+ auto seriesB = Generate(seedB, count);
+
+ std::vector<float> output;
+ for (int i = 0; i < count; i++)
+ output.push_back(seriesA[i] * (1 - crossSeed) + seriesB[i] * crossSeed);
+
+ return output;
+ }
+}
diff --git a/cbb_RoomReverb/reverb_utils/RandomBuffer.h b/cbb_RoomReverb/reverb_utils/RandomBuffer.h
new file mode 100644
index 0000000..8692368
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/RandomBuffer.h
@@ -0,0 +1,15 @@
+
+#pragma once
+
+#include <vector>
+#include <stdint.h>
+
+namespace ReverbHallRoom
+{
+ class RandomBuffer
+ {
+ public:
+ static std::vector<float> Generate(uint64_t seed, int count);
+ static std::vector<float> Generate(uint64_t seed, int count, float crossSeed);
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/ReverbChannel.h b/cbb_RoomReverb/reverb_utils/ReverbChannel.h
new file mode 100644
index 0000000..12ff43a
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/ReverbChannel.h
@@ -0,0 +1,407 @@
+
+#pragma once
+
+#include <map>
+#include <memory>
+#include "Parameters.h"
+#include "ModulatedDelay.h"
+#include "MultitapDelay.h"
+#include "RandomBuffer.h"
+#include "Lp1.h"
+#include "Hp1.h"
+#include "DelayLine.h"
+#include "AllpassDiffuser.h"
+#include <cmath>
+#include "ReverbChannel.h"
+#include "Utils.h"
+
+namespace ReverbHallRoom
+{
+ enum class ChannelLR
+ {
+ Left,
+ Right
+ };
+
+ class ReverbChannel
+ {
+ private:
+ static const int TotalLineCount = 12;
+
+ double paramsScaled[Parameter::COUNT] = { 0.0 };
+ int samplerate;
+
+ ModulatedDelay preDelay;
+ MultitapDelay multitap;
+ AllpassDiffuser diffuser;
+ DelayLine lines[TotalLineCount];
+ RandomBuffer rand;
+ Hp1 highPass;
+ Lp1 lowPass;
+
+ int delayLineSeed;
+ int postDiffusionSeed;
+
+ // Used the the main process loop
+ int lineCount;
+
+ bool lowCutEnabled;
+ bool highCutEnabled;
+ bool multitapEnabled;
+ bool diffuserEnabled;
+ float inputMix;
+ float dryOut;
+ float earlyOut;
+ float lineOut;
+ float crossSeed;
+ ChannelLR channelLr;
+
+ public:
+
+ ReverbChannel(int samplerate, ChannelLR leftOrRight)
+ {
+ this->channelLr = leftOrRight;
+ crossSeed = 0.0;
+ lineCount = 8;
+ diffuser.SetInterpolationEnabled(true);
+ highPass.SetCutoffHz(20);
+ lowPass.SetCutoffHz(20000);
+ SetSamplerate(samplerate);
+ }
+
+ int GetSamplerate()
+ {
+ return samplerate;
+ }
+
+ void SetSamplerate(int samplerate)
+ {
+ this->samplerate = samplerate;
+ highPass.SetSamplerate(samplerate);
+ lowPass.SetSamplerate(samplerate);
+ diffuser.SetSamplerate(samplerate);
+
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetSamplerate(samplerate);
+
+ ReapplyAllParams();
+ ClearBuffers();
+ UpdateLines();
+ }
+
+ void ReapplyAllParams()
+ {
+ for (int i = 0; i < Parameter::COUNT; i++)
+ SetParameter(i, paramsScaled[i]);
+ }
+
+ void SetParameter(int para, double scaledValue)
+ {
+ paramsScaled[para] = scaledValue;
+
+ switch (para)
+ {
+ case Parameter::Interpolation:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetInterpolationEnabled(scaledValue >= 0.5);
+ break;
+ case Parameter::LowCutEnabled:
+ lowCutEnabled = scaledValue >= 0.5;
+ if (lowCutEnabled)
+ highPass.ClearBuffers();
+ break;
+ case Parameter::HighCutEnabled:
+ highCutEnabled = scaledValue >= 0.5;
+ if (highCutEnabled)
+ lowPass.ClearBuffers();
+ break;
+ case Parameter::InputMix:
+ inputMix = scaledValue;
+ break;
+ case Parameter::LowCut:
+ highPass.SetCutoffHz(scaledValue);
+ break;
+ case Parameter::HighCut:
+ lowPass.SetCutoffHz(scaledValue);
+ break;
+ case Parameter::DryOut:
+ dryOut = scaledValue <= -30 ? 0.0 : Utils::DB2Gainf(scaledValue);
+ break;
+ case Parameter::EarlyOut:
+ earlyOut = scaledValue <= -30 ? 0.0 : Utils::DB2Gainf(scaledValue);
+ break;
+ case Parameter::LateOut:
+ lineOut = scaledValue <= -30 ? 0.0 : Utils::DB2Gainf(scaledValue);
+ break;
+
+
+ case Parameter::TapEnabled:
+ {
+ auto newVal = scaledValue >= 0.5;
+ if (newVal != multitapEnabled)
+ multitap.ClearBuffers();
+ multitapEnabled = newVal;
+ break;
+ }
+ case Parameter::TapCount:
+ multitap.SetTapCount((int)scaledValue);
+ break;
+ case Parameter::TapDecay:
+ multitap.SetTapDecay(scaledValue);
+ break;
+ case Parameter::TapPredelay:
+ preDelay.SampleDelay = (int)Ms2Samples(scaledValue);
+ break;
+ case Parameter::TapLength:
+ multitap.SetTapLength((int)Ms2Samples(scaledValue));
+ break;
+
+
+ case Parameter::EarlyDiffuseEnabled:
+ {
+ auto newVal = scaledValue >= 0.5;
+ if (newVal != diffuserEnabled)
+ diffuser.ClearBuffers();
+ diffuserEnabled = newVal;
+ break;
+ }
+ case Parameter::EarlyDiffuseCount:
+ diffuser.Stages = (int)scaledValue;
+ break;
+ case Parameter::EarlyDiffuseDelay:
+ diffuser.SetDelay((int)Ms2Samples(scaledValue));
+ break;
+ case Parameter::EarlyDiffuseModAmount:
+ diffuser.SetModulationEnabled(scaledValue > 0.5);
+ diffuser.SetModAmount(Ms2Samples(scaledValue));
+ break;
+ case Parameter::EarlyDiffuseFeedback:
+ diffuser.SetFeedback(scaledValue);
+ break;
+ case Parameter::EarlyDiffuseModRate:
+ diffuser.SetModRate(scaledValue);
+ break;
+
+
+ case Parameter::LateMode:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].TapPostDiffuser = scaledValue >= 0.5;
+ break;
+ case Parameter::LateLineCount:
+ lineCount = (int)scaledValue;
+ break;
+ case Parameter::LateDiffuseEnabled:
+ for (int i = 0; i < TotalLineCount; i++)
+ {
+ auto newVal = scaledValue >= 0.5;
+ if (newVal != lines[i].DiffuserEnabled)
+ lines[i].ClearDiffuserBuffer();
+ lines[i].DiffuserEnabled = newVal;
+ }
+ break;
+ case Parameter::LateDiffuseCount:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetDiffuserStages((int)scaledValue);
+ break;
+ case Parameter::LateLineSize:
+ UpdateLines();
+ break;
+ case Parameter::LateLineModAmount:
+ UpdateLines();
+ break;
+ case Parameter::LateDiffuseDelay:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetDiffuserDelay((int)Ms2Samples(scaledValue));
+ break;
+ case Parameter::LateDiffuseModAmount:
+ UpdateLines();
+ break;
+ case Parameter::LateLineDecay:
+ UpdateLines();
+ break;
+ case Parameter::LateLineModRate:
+ UpdateLines();
+ break;
+ case Parameter::LateDiffuseFeedback:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetDiffuserFeedback(scaledValue);
+ break;
+ case Parameter::LateDiffuseModRate:
+ UpdateLines();
+ break;
+
+
+ case Parameter::EqLowShelfEnabled:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].LowShelfEnabled = scaledValue >= 0.5;
+ break;
+ case Parameter::EqHighShelfEnabled:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].HighShelfEnabled = scaledValue >= 0.5;
+ break;
+ case Parameter::EqLowpassEnabled:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].CutoffEnabled = scaledValue >= 0.5;
+ break;
+ case Parameter::EqLowFreq:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetLowShelfFrequency(scaledValue);
+ break;
+ case Parameter::EqHighFreq:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetHighShelfFrequency(scaledValue);
+ break;
+ case Parameter::EqCutoff:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetCutoffFrequency(scaledValue);
+ break;
+ case Parameter::EqLowGain:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetLowShelfGain(scaledValue);
+ break;
+ case Parameter::EqHighGain:
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetHighShelfGain(scaledValue);
+ break;
+
+
+ case Parameter::EqCrossSeed:
+ crossSeed = channelLr == ChannelLR::Right ? 0.5 * scaledValue : 1 - 0.5 * scaledValue;
+ multitap.SetCrossSeed(crossSeed);
+ diffuser.SetCrossSeed(crossSeed);
+ UpdateLines();
+ UpdatePostDiffusion();
+ break;
+
+
+ case Parameter::SeedTap:
+ multitap.SetSeed((int)scaledValue);
+ break;
+ case Parameter::SeedDiffusion:
+ diffuser.SetSeed((int)scaledValue);
+ break;
+ case Parameter::SeedDelay:
+ delayLineSeed = (int)scaledValue;
+ UpdateLines();
+ break;
+ case Parameter::SeedPostDiffusion:
+ postDiffusionSeed = (int)scaledValue;
+ UpdatePostDiffusion();
+ break;
+ }
+ }
+
+ void Process(float* input, float* output, int bufSize)
+ {
+ float tempBuffer[BUFFER_SIZE];
+ float earlyOutBuffer[BUFFER_SIZE];
+ float lineOutBuffer[BUFFER_SIZE];
+ float lineSumBuffer[BUFFER_SIZE];
+
+ Utils::Copy(tempBuffer, input, bufSize);
+
+ if (lowCutEnabled)
+ highPass.Process(tempBuffer, tempBuffer, bufSize);
+ if (highCutEnabled)
+ lowPass.Process(tempBuffer, tempBuffer, bufSize);
+
+ // completely zero if no input present
+ // Previously, the very small values were causing some really strange CPU spikes
+ for (int i = 0; i < bufSize; i++)
+ {
+ auto n = tempBuffer[i];
+ if (n * n < 0.000000001)
+ tempBuffer[i] = 0;
+ }
+
+ preDelay.Process(tempBuffer, tempBuffer, bufSize);
+ if (multitapEnabled)
+ multitap.Process(tempBuffer, tempBuffer, bufSize);
+ if (diffuserEnabled)
+ diffuser.Process(tempBuffer, tempBuffer, bufSize);
+
+ Utils::Copy(earlyOutBuffer, tempBuffer, bufSize);
+ Utils::ZeroBuffer(lineSumBuffer, bufSize);
+ for (int i = 0; i < lineCount; i++)
+ {
+ lines[i].Process(tempBuffer, lineOutBuffer, bufSize);
+ Utils::Mix(lineSumBuffer, lineOutBuffer, 1.0f, bufSize);
+ }
+
+ auto perLineGain = GetPerLineGain();
+ Utils::Gain(lineSumBuffer, perLineGain, bufSize);
+
+ for (int i = 0; i < bufSize; i++)
+ {
+ output[i] = dryOut * input[i]
+ + earlyOut * earlyOutBuffer[i]
+ + lineOut * lineSumBuffer[i];
+ }
+ }
+
+ void ClearBuffers()
+ {
+ lowPass.ClearBuffers();
+ highPass.ClearBuffers();
+ preDelay.ClearBuffers();
+ multitap.ClearBuffers();
+ diffuser.ClearBuffers();
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].ClearBuffers();
+ }
+
+
+ private:
+ float GetPerLineGain()
+ {
+ return 1.0 / std::sqrt(lineCount);
+ }
+
+ void UpdateLines()
+ {
+ auto lineDelaySamples = (int)Ms2Samples(paramsScaled[Parameter::LateLineSize]);
+ auto lineDecayMillis = paramsScaled[Parameter::LateLineDecay] * 1000;
+ auto lineDecaySamples = Ms2Samples(lineDecayMillis);
+
+ auto lineModAmount = Ms2Samples(paramsScaled[Parameter::LateLineModAmount]);
+ auto lineModRate = paramsScaled[Parameter::LateLineModRate];
+
+ auto lateDiffusionModAmount = Ms2Samples(paramsScaled[Parameter::LateDiffuseModAmount]);
+ auto lateDiffusionModRate = paramsScaled[Parameter::LateDiffuseModRate];
+
+ auto delayLineSeeds = RandomBuffer::Generate(delayLineSeed, TotalLineCount * 3, crossSeed);
+
+ for (int i = 0; i < TotalLineCount; i++)
+ {
+ auto modAmount = lineModAmount * (0.7 + 0.3 * delayLineSeeds[i]);
+ auto modRate = lineModRate * (0.7 + 0.3 * delayLineSeeds[TotalLineCount + i]) / samplerate;
+
+ auto delaySamples = (0.5 + 1.0 * delayLineSeeds[TotalLineCount * 2 + i]) * lineDelaySamples;
+ if (delaySamples < modAmount + 2) // when the delay is set really short, and the modulation is very high
+ delaySamples = modAmount + 2; // the mod could actually take the delay time negative, prevent that! -- provide 2 extra sample as margin of safety
+
+ auto dbAfter1Iteration = delaySamples / lineDecaySamples * (-60); // lineDecay is the time it takes to reach T60
+ auto gainAfter1Iteration = Utils::DB2Gainf(dbAfter1Iteration);
+
+ lines[i].SetDelay((int)delaySamples);
+ lines[i].SetFeedback(gainAfter1Iteration);
+ lines[i].SetLineModAmount(modAmount);
+ lines[i].SetLineModRate(modRate);
+ lines[i].SetDiffuserModAmount(lateDiffusionModAmount);
+ lines[i].SetDiffuserModRate(lateDiffusionModRate);
+ }
+ }
+
+ void UpdatePostDiffusion()
+ {
+ for (int i = 0; i < TotalLineCount; i++)
+ lines[i].SetDiffuserSeed((postDiffusionSeed) * (i + 1), crossSeed);
+ }
+
+ float Ms2Samples(float value)
+ {
+ return value / 1000.0f * samplerate;
+ }
+
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/ReverbController.h b/cbb_RoomReverb/reverb_utils/ReverbController.h
new file mode 100644
index 0000000..3d0c061
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/ReverbController.h
@@ -0,0 +1,106 @@
+
+#pragma once
+
+#include <stdio.h>
+#include <vector>
+#include "Parameters.h"
+#include "ReverbChannel.h"
+#include "AllpassDiffuser.h"
+#include "MultitapDelay.h"
+#include "Utils.h"
+
+namespace ReverbHallRoom
+{
+ class ReverbController
+ {
+ private:
+ int samplerate;
+
+ ReverbChannel channelL;
+ ReverbChannel channelR;
+ double parameters[(int)Parameter::COUNT] = {0};
+
+ public:
+ ReverbController(int samplerate = 48000) :
+ channelL(samplerate, ChannelLR::Left),
+ channelR(samplerate, ChannelLR::Right)
+ {
+ this->samplerate = samplerate;
+ }
+
+ int GetSamplerate()
+ {
+ return samplerate;
+ }
+
+ void SetSamplerate(int samplerate)
+ {
+ this->samplerate = samplerate;
+ channelL.SetSamplerate(samplerate);
+ channelR.SetSamplerate(samplerate);
+ }
+
+ int GetParameterCount()
+ {
+ return Parameter::COUNT;
+ }
+
+ double* GetAllParameters()
+ {
+ return parameters;
+ }
+
+ void SetParameter(int paramId, double value)
+ {
+ parameters[paramId] = value;
+ auto scaled = ScaleParam(value, paramId);
+ channelL.SetParameter(paramId, scaled);
+ channelR.SetParameter(paramId, scaled);
+ }
+
+ void ClearBuffers()
+ {
+ channelL.ClearBuffers();
+ channelR.ClearBuffers();
+ }
+
+ void Process(float* inL, float* inR, float* outL, float* outR, int bufSize)
+ {
+ float outLTemp[BUFFER_SIZE];
+ float outRTemp[BUFFER_SIZE];
+
+ while (bufSize > 0)
+ {
+ int subBufSize = bufSize > BUFFER_SIZE ? BUFFER_SIZE : bufSize;
+ ProcessChunk(inL, inR, outLTemp, outRTemp, subBufSize);
+ Utils::Copy(outL, outLTemp, subBufSize);
+ Utils::Copy(outR, outRTemp, subBufSize);
+ inL = &inL[subBufSize];
+ inR = &inR[subBufSize];
+ outL = &outL[subBufSize];
+ outR = &outR[subBufSize];
+ bufSize -= subBufSize;
+ }
+ }
+
+ private:
+ void ProcessChunk(float* inL, float* inR, float* outL, float* outR, int bufSize)
+ {
+ float leftChannelIn[BUFFER_SIZE];
+ float rightChannelIn[BUFFER_SIZE];
+
+ float inputMix = ScaleParam(parameters[Parameter::InputMix], Parameter::InputMix);
+ float cm = inputMix * 0.5;
+ float cmi = (1 - cm);
+
+ for (int i = 0; i < bufSize; i++)
+ {
+ leftChannelIn[i] = inL[i] * cmi + inR[i] * cm;
+ rightChannelIn[i] = inR[i] * cmi + inL[i] * cm;
+ }
+
+ channelL.Process(leftChannelIn, outL, bufSize);
+ channelR.Process(rightChannelIn, outR, bufSize);
+ }
+ };
+}
diff --git a/cbb_RoomReverb/reverb_utils/Utils.h b/cbb_RoomReverb/reverb_utils/Utils.h
new file mode 100644
index 0000000..2ec7a63
--- /dev/null
+++ b/cbb_RoomReverb/reverb_utils/Utils.h
@@ -0,0 +1,91 @@
+
+#pragma once
+
+#define _USE_MATH_DEFINES 1
+#include <math.h>
+#include <stdint.h>
+
+namespace ReverbHallRoom
+{
+ namespace Utils
+ {
+ template<typename T>
+ inline void ZeroBuffer(T* buffer, int len)
+ {
+ for (int i = 0; i < len; i++)
+ buffer[i] = 0;
+ }
+
+ template<typename T>
+ inline void Copy(T* dest, T* source, int len)
+ {
+ memcpy(dest, source, len * sizeof(T));
+ }
+
+ template<typename T>
+ inline void Gain(T* buffer, T gain, int len)
+ {
+ for (int i = 0; i < len; i++)
+ {
+ buffer[i] *= gain;
+ }
+ }
+
+ template<typename T>
+ inline void Mix(T* target, T* source, T gain, int len)
+ {
+ for (int i = 0; i < len; i++)
+ target[i] += source[i] * gain;
+ }
+
+ inline float DB2Gainf(float input)
+ {
+ //return std::pow(10.0f, input / 20.0f);
+ return powf(10, input * 0.05f);
+ }
+
+ template<typename T>
+ inline double DB2Gain(T input)
+ {
+ return pow10f(input / 20.0);
+ }
+
+ template<typename T>
+ inline double Gain2DB(T input)
+ {
+ //if (input < 0.0000001)
+ // return -100000;
+
+ return 20.0f * log10f(input);
+ }
+
+ const float dec1Mult = (10 / 9.0) * 0.1;
+ const float dec2Mult = (100 / 99.0) * 0.01;
+ const float dec3Mult = (1000 / 999.0) * 0.001;
+ const float dec4Mult = (10000 / 9999.0) * 0.0001;
+
+ const float oct1Mult = (2 / 1.0) * 0.5;
+ const float oct2Mult = (4 / 3.0) * 0.25;
+ const float oct3Mult = (8 / 7.0) * 0.125;
+ const float oct4Mult = (16 / 15.0) * 0.0625;
+ const float oct5Mult = (32 / 31.0) * 0.03125;
+ const float oct6Mult = (64 / 63.0) * 0.015625;
+ const float oct7Mult = (128 / 127.0) * 0.0078125;
+ const float oct8Mult = (256 / 255.0) * 0.00390625;
+
+ inline float Resp1dec(float x) { return (powf(10, x) - 1) * dec1Mult; }
+ inline float Resp2dec(float x) { return (powf(10, 2 * x) - 1) * dec2Mult; }
+ inline float Resp3dec(float x) { return (powf(10, 3 * x) - 1) * dec3Mult; }
+ inline float Resp4dec(float x) { return (powf(10, 4 * x) - 1) * dec4Mult; }
+
+ inline float Resp1oct(float x) { return (powf(2, x) - 1) * oct1Mult; }
+ inline float Resp2oct(float x) { return (powf(2, 2 * x) - 1) * oct2Mult; }
+ inline float Resp3oct(float x) { return (powf(2, 3 * x) - 1) * oct3Mult; }
+ inline float Resp4oct(float x) { return (powf(2, 4 * x) - 1) * oct4Mult; }
+ inline float Resp5oct(float x) { return (powf(2, 5 * x) - 1) * oct5Mult; }
+ inline float Resp6oct(float x) { return (powf(2, 6 * x) - 1) * oct6Mult; }
+ inline float Resp7oct(float x) { return (powf(2, 7 * x) - 1) * oct7Mult; }
+ inline float Resp8oct(float x) { return (powf(2, 8 * x) - 1) * oct8Mult; }
+
+ }
+}
diff --git a/cbb_RoomReverb/reverb_wrapper.c b/cbb_RoomReverb/reverb_wrapper.c
new file mode 100644
index 0000000..a184c54
--- /dev/null
+++ b/cbb_RoomReverb/reverb_wrapper.c
@@ -0,0 +1,73 @@
+#include "reverb.h"
+//#include "reverb_wrapper.h"
+#include <stdio.h>
+
+
+static int ch,n,L;
+static float *x[32], *y[32];
+
+
+void reverb_wrapper_init(void **p, int channels, int frame_size, int sample_rate, double dry, double early, double late, int input_mix_on, int hight_cut_on, // 6
+ int low_cut_on, double input_mix, double high_cut_freq, double low_cut_freq, double cross_seed, int taps_on, double taps_count, double taps_pre_delay, double taps_decay, double taps_length, int early_difus_on, double early_difus_count, double early_difus_delay, // 13
+ double early_difus_feedback, double early_difus_mod_amt, double early_difus_mod_rate, int late_mode, int late_reflect_on, double late_line_count, double late_line_size, double late_line_mod_amt, double late_line_decay, double late_line_mod_rate, double late_difus_count, double late_difus_delay, // 12
+ double late_difus_feedback, double late_difus_mod_amt, double late_difus_mod_rate, int eq_low_shelf_on, int eq_high_shelf_on, int eq_low_pass_on, double eq_low_shelf_freq, double eq_low_shelf_gain, double eq_high_shelf_freq, double eq_high_shelf_gain, double eq_low_pass_freq // 11
+ )
+{
+ Reverb_common c_t = {sample_rate, dry, early, late, input_mix_on, hight_cut_on, low_cut_on, input_mix, high_cut_freq, low_cut_freq, cross_seed};
+ Reverb_taps t_t = {taps_on, taps_count, taps_pre_delay, taps_decay, taps_length};
+ Reverb_early e_t = {early_difus_on, early_difus_count, early_difus_delay, early_difus_feedback, early_difus_mod_amt, early_difus_mod_rate};
+ Reverb_late l_t = {late_mode, late_reflect_on, late_line_count, late_line_size, late_line_mod_amt, late_line_decay, late_line_mod_rate, late_difus_count, late_difus_delay, late_difus_feedback, late_difus_mod_amt, late_difus_mod_rate};
+ Reverb_eq eq_t = {eq_low_shelf_on, eq_high_shelf_on, eq_low_pass_on, eq_low_shelf_freq, eq_low_shelf_gain, eq_high_shelf_freq, eq_high_shelf_gain, eq_low_pass_freq};
+
+ *(Reverb **)p = reverb_new(channels, frame_size, &c_t, &t_t, &e_t, &l_t, &eq_t);
+ ch = channels;
+ n = frame_size;
+
+// FILE *file = fopen("debug.txt", "w");
+// if (NULL == file) return;
+
+// fprintf(file, "GUI params:\n");
+// fprintf(file, "late_refl_on:%d.\n", l_t.late_reflect_on);
+// fprintf(file, "FS:%d dry:%2.2f early:%2.2f late:%2.2f inp_on:%d h_c_on:%d l_c_on:%d in_mix:%2.2f h_c:%2.2f l_c:%2.2f c_s:%2.2f.\n",
+// c_t.sample_rate, c_t.dry, c_t.early, c_t.late, c_t.input_mix_on, c_t.high_cut_on, \
+// c_t.low_cut_on, c_t.input_mix, c_t.high_cut, c_t.low_cut, c_t.cross_seed);
+// fclose(file);
+}
+
+
+void reverb_wrapper_process(void *p, /*int sample_rate, double dry, double early, double late, int input_mix_on, int hight_cut_on, // 6
+ int low_cut_on, double input_mix, double high_cut_freq, double low_cut_freq, double cross_seed, int taps_on, double taps_count, double taps_pre_delay, double taps_decay, double taps_length, int early_difus_on, double early_difus_count, double early_difus_delay, // 13
+ double early_difus_feedback, double early_difus_mod_amt, double early_difus_mod_rate, int late_mode, int late_reflect_on, double late_line_count, double late_line_size, double late_line_mod_amt, double late_line_decay, double late_line_mod_rate, double late_difus_count, double late_difus_delay, // 12
+ double late_difus_feedback, double late_difus_mod_amt, double late_difus_mod_rate, int eq_low_shelf_on, int eq_high_shelf_on, int eq_low_pass_on, double eq_low_shelf_freq, double eq_low_shelf_gain, double eq_high_shelf_freq, double eq_high_shelf_gain, double eq_low_pass_freq,
+ */float *src, float *dst)
+{
+ int i;
+
+ for(i = 0; i < ch; i ++)
+ x[i] = src + i * n;
+
+ for(i = 0; i < ch; i ++)
+ y[i] = dst + i * n;
+
+
+ reverb_process(p, x, y);
+
+ /*Reverb params = {ch, n, {sample_rate, dry, early, late, input_mix_on, hight_cut_on, low_cut_on, input_mix, high_cut_freq, low_cut_freq, cross_seed},
+ {taps_on, taps_count, taps_pre_delay, taps_decay, taps_length},
+ {early_difus_on, early_difus_count, early_difus_delay, early_difus_feedback, early_difus_mod_amt, early_difus_mod_rate},
+ {late_mode, late_reflect_on, late_line_count, late_line_size, late_line_mod_amt, late_line_mod_rate},
+ {eq_low_shelf_on, eq_high_shelf_on, eq_low_pass_on, eq_low_shelf_freq, eq_low_shelf_gain, eq_high_shelf_freq, eq_high_shelf_gain, eq_low_pass_freq},
+ };
+ reverb_params_set(¶ms);*/
+
+}
+
+
+void reverb_wrapper_delete(void *p)
+{
+ reverb_delete(p);
+}
+
+
+
+
diff --git a/cbb_RoomReverb/reverb_wrapper.h b/cbb_RoomReverb/reverb_wrapper.h
new file mode 100644
index 0000000..f58ef7e
--- /dev/null
+++ b/cbb_RoomReverb/reverb_wrapper.h
@@ -0,0 +1,57 @@
+#ifndef REVERB_WRAPPER_H
+#define REVERB_WRAPPER_H
+
+
+#define UI_FULL 0
+#define UI_XYK 1
+#define GUI UI_FULL
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+#if (GUI == UI_XYK)
+
+ void reverb_wrapper_init(void **p, int channels, int frame_size, int sample_rate, double dry, double early, double late, int input_mix_on, int hight_cut_on, // 1~6
+ int low_cut_on, double in_mix, double high_cut_freq, double low_cut_freq, double cross_seed, int taps_on, double taps_count, double taps_pre_delay, double taps_decay, double taps_length, int early_difus_on, double early_difus_count, double early_difus_delay, // 7~19
+ double early_difus_feedback, double early_difus_mod_amt, double early_difus_mod_rate, int late_mode, int late_reflect_on, double late_line_count, double late_line_size, double late_line_mod_amt, double late_line_decay, double late_line_mod_rate, double late_difus_count, double late_difus_delay, // 20~31
+ double late_difus_feedback, double late_difus_mod_amt, double late_difus_mod_rate, int eq_low_shelf_on, int eq_high_shelf_on, int eq_low_pass_on, double eq_low_shelf_freq, double eq_low_shelf_gain, double eq_high_shelf_freq, double eq_high_shelf_gain, double eq_low_pass_freq //32~42
+ );
+
+ void reverb_wrapper_process(void *p, int sample_rate, double dry, double early, double late, int input_mix_on, int hight_cut_on, // 6
+ int low_cut_on, double input_mix, double high_cut_freq, double low_cut_freq, double cross_seed, int taps_on, double taps_count, double taps_pre_delay, double taps_decay, double taps_length, int early_difus_on, double early_difus_count, double early_difus_delay, // 13
+ double early_difus_feedback, double early_difus_mod_amt, double early_difus_mod_rate, int late_mode, int late_reflect_on, double late_line_count, double late_line_size, double late_line_mod_amt, double late_line_decay, double late_line_mod_rate, double late_difus_count, double late_difus_delay, // 12
+ double late_difus_feedback, double late_difus_mod_amt, double late_difus_mod_rate, int eq_low_shelf_on, int eq_high_shelf_on, int eq_low_pass_on, double eq_low_shelf_freq, double eq_low_shelf_gain, double eq_high_shelf_freq, double eq_high_shelf_gain, double eq_low_pass_freq,
+ float *src, float *dst);
+
+#else
+
+ void reverb_wrapper_init(void **p, int channels, int frame_size, int sample_rate, double dry, double early, double late, int input_mix_on, int hight_cut_on, // 1~6
+ int low_cut_on, double in_mix, double high_cut_freq, double low_cut_freq, double cross_seed, int taps_on, double taps_count, double taps_pre_delay, double taps_decay, double taps_length, int early_difus_on, double early_difus_count, double early_difus_delay, // 7~19
+ double early_difus_feedback, double early_difus_mod_amt, double early_difus_mod_rate, int late_mode, int late_reflect_on, double late_line_count, double late_line_size, double late_line_mod_amt, double late_line_decay, double late_line_mod_rate, double late_difus_count, double late_difus_delay, // 20~31
+ double late_difus_feedback, double late_difus_mod_amt, double late_difus_mod_rate, int eq_low_shelf_on, int eq_high_shelf_on, int eq_low_pass_on, double eq_low_shelf_freq, double eq_low_shelf_gain, double eq_high_shelf_freq, double eq_high_shelf_gain, double eq_low_pass_freq //32~42
+ );
+
+ void reverb_wrapper_process(void *p, /*int sample_rate, double dry, double early, double late, int input_mix_on, int hight_cut_on, // 6
+ int low_cut_on, double input_mix, double high_cut_freq, double low_cut_freq, double cross_seed, int taps_on, double taps_count, double taps_pre_delay, double taps_decay, double taps_length, int early_difus_on, double early_difus_count, double early_difus_delay, // 13
+ double early_difus_feedback, double early_difus_mod_amt, double early_difus_mod_rate, int late_mode, int late_reflect_on, double late_line_count, double late_line_size, double late_line_mod_amt, double late_line_decay, double late_line_mod_rate, double late_difus_count, double late_difus_delay, // 12
+ double late_difus_feedback, double late_difus_mod_amt, double late_difus_mod_rate, int eq_low_shelf_on, int eq_high_shelf_on, int eq_low_pass_on, double eq_low_shelf_freq, double eq_low_shelf_gain, double eq_high_shelf_freq, double eq_high_shelf_gain, double eq_low_pass_freq,
+ */float *src, float *dst);
+
+#endif
+
+void reverb_wrapper_delete(void *p);
+
+
+
+
+#endif
+
+
+
+
+#ifdef __cplusplus
+}
+#endif
\ No newline at end of file
--
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