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-rw-r--r--dom/media/AudioConverter.cpp393
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diff --git a/dom/media/AudioConverter.cpp b/dom/media/AudioConverter.cpp
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+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioConverter.h"
+#include <string.h>
+#include <speex/speex_resampler.h>
+#include <cmath>
+
+/*
+ * Parts derived from MythTV AudioConvert Class
+ * Created by Jean-Yves Avenard.
+ *
+ * Copyright (C) Bubblestuff Pty Ltd 2013
+ * Copyright (C) foobum@gmail.com 2010
+ */
+
+namespace mozilla {
+
+AudioConverter::AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut)
+ : mIn(aIn)
+ , mOut(aOut)
+ , mResampler(nullptr)
+{
+ MOZ_DIAGNOSTIC_ASSERT(aIn.Format() == aOut.Format() &&
+ aIn.Interleaved() == aOut.Interleaved(),
+ "No format or rate conversion is supported at this stage");
+ MOZ_DIAGNOSTIC_ASSERT(aOut.Channels() <= 2 ||
+ aIn.Channels() == aOut.Channels(),
+ "Only down/upmixing to mono or stereo is supported at this stage");
+ MOZ_DIAGNOSTIC_ASSERT(aOut.Interleaved(), "planar audio format not supported");
+ mIn.Layout().MappingTable(mOut.Layout(), mChannelOrderMap);
+ if (aIn.Rate() != aOut.Rate()) {
+ RecreateResampler();
+ }
+}
+
+AudioConverter::~AudioConverter()
+{
+ if (mResampler) {
+ speex_resampler_destroy(mResampler);
+ mResampler = nullptr;
+ }
+}
+
+bool
+AudioConverter::CanWorkInPlace() const
+{
+ bool needDownmix = mIn.Channels() > mOut.Channels();
+ bool needUpmix = mIn.Channels() < mOut.Channels();
+ bool canDownmixInPlace =
+ mIn.Channels() * AudioConfig::SampleSize(mIn.Format()) >=
+ mOut.Channels() * AudioConfig::SampleSize(mOut.Format());
+ bool needResample = mIn.Rate() != mOut.Rate();
+ bool canResampleInPlace = mIn.Rate() >= mOut.Rate();
+ // We should be able to work in place if 1s of audio input takes less space
+ // than 1s of audio output. However, as we downmix before resampling we can't
+ // perform any upsampling in place (e.g. if incoming rate >= outgoing rate)
+ return !needUpmix && (!needDownmix || canDownmixInPlace) &&
+ (!needResample || canResampleInPlace);
+}
+
+size_t
+AudioConverter::ProcessInternal(void* aOut, const void* aIn, size_t aFrames)
+{
+ if (mIn.Channels() > mOut.Channels()) {
+ return DownmixAudio(aOut, aIn, aFrames);
+ } else if (mIn.Channels() < mOut.Channels()) {
+ return UpmixAudio(aOut, aIn, aFrames);
+ } else if (mIn.Layout() != mOut.Layout() && CanReorderAudio()) {
+ ReOrderInterleavedChannels(aOut, aIn, aFrames);
+ } else if (aIn != aOut) {
+ memmove(aOut, aIn, FramesOutToBytes(aFrames));
+ }
+ return aFrames;
+}
+
+// Reorder interleaved channels.
+// Can work in place (e.g aOut == aIn).
+template <class AudioDataType>
+void
+_ReOrderInterleavedChannels(AudioDataType* aOut, const AudioDataType* aIn,
+ uint32_t aFrames, uint32_t aChannels,
+ const uint8_t* aChannelOrderMap)
+{
+ MOZ_DIAGNOSTIC_ASSERT(aChannels <= MAX_AUDIO_CHANNELS);
+ AudioDataType val[MAX_AUDIO_CHANNELS];
+ for (uint32_t i = 0; i < aFrames; i++) {
+ for (uint32_t j = 0; j < aChannels; j++) {
+ val[j] = aIn[aChannelOrderMap[j]];
+ }
+ for (uint32_t j = 0; j < aChannels; j++) {
+ aOut[j] = val[j];
+ }
+ aOut += aChannels;
+ aIn += aChannels;
+ }
+}
+
+void
+AudioConverter::ReOrderInterleavedChannels(void* aOut, const void* aIn,
+ size_t aFrames) const
+{
+ MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() == mOut.Channels());
+
+ if (mOut.Channels() == 1 || mOut.Layout() == mIn.Layout()) {
+ // If channel count is 1, planar and non-planar formats are the same and
+ // there's nothing to reorder.
+ if (aOut != aIn) {
+ memmove(aOut, aIn, FramesOutToBytes(aFrames));
+ }
+ return;
+ }
+
+ uint32_t bits = AudioConfig::FormatToBits(mOut.Format());
+ switch (bits) {
+ case 8:
+ _ReOrderInterleavedChannels((uint8_t*)aOut, (const uint8_t*)aIn,
+ aFrames, mIn.Channels(), mChannelOrderMap);
+ break;
+ case 16:
+ _ReOrderInterleavedChannels((int16_t*)aOut,(const int16_t*)aIn,
+ aFrames, mIn.Channels(), mChannelOrderMap);
+ break;
+ default:
+ MOZ_DIAGNOSTIC_ASSERT(AudioConfig::SampleSize(mOut.Format()) == 4);
+ _ReOrderInterleavedChannels((int32_t*)aOut,(const int32_t*)aIn,
+ aFrames, mIn.Channels(), mChannelOrderMap);
+ break;
+ }
+}
+
+static inline int16_t clipTo15(int32_t aX)
+{
+ return aX < -32768 ? -32768 : aX <= 32767 ? aX : 32767;
+}
+
+size_t
+AudioConverter::DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const
+{
+ MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
+ mIn.Format() == AudioConfig::FORMAT_FLT);
+ MOZ_ASSERT(mIn.Channels() >= mOut.Channels());
+ MOZ_ASSERT(mIn.Layout() == AudioConfig::ChannelLayout(mIn.Channels()),
+ "Can only downmix input data in SMPTE layout");
+ MOZ_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
+ mOut.Layout() == AudioConfig::ChannelLayout(1));
+
+ uint32_t channels = mIn.Channels();
+
+ if (channels == 1 && mOut.Channels() == 1) {
+ if (aOut != aIn) {
+ memmove(aOut, aIn, FramesOutToBytes(aFrames));
+ }
+ return aFrames;
+ }
+
+ if (channels > 2) {
+ if (mIn.Format() == AudioConfig::FORMAT_FLT) {
+ // Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
+ static const float dmatrix[6][8][2]= {
+ /*3*/{{0.5858f,0},{0,0.5858f},{0.4142f,0.4142f}},
+ /*4*/{{0.4226f,0},{0,0.4226f},{0.366f, 0.2114f},{0.2114f,0.366f}},
+ /*5*/{{0.6510f,0},{0,0.6510f},{0.4600f,0.4600f},{0.5636f,0.3254f},{0.3254f,0.5636f}},
+ /*6*/{{0.5290f,0},{0,0.5290f},{0.3741f,0.3741f},{0.3741f,0.3741f},{0.4582f,0.2645f},{0.2645f,0.4582f}},
+ /*7*/{{0.4553f,0},{0,0.4553f},{0.3220f,0.3220f},{0.3220f,0.3220f},{0.2788f,0.2788f},{0.3943f,0.2277f},{0.2277f,0.3943f}},
+ /*8*/{{0.3886f,0},{0,0.3886f},{0.2748f,0.2748f},{0.2748f,0.2748f},{0.3366f,0.1943f},{0.1943f,0.3366f},{0.3366f,0.1943f},{0.1943f,0.3366f}},
+ };
+ // Re-write the buffer with downmixed data
+ const float* in = static_cast<const float*>(aIn);
+ float* out = static_cast<float*>(aOut);
+ for (uint32_t i = 0; i < aFrames; i++) {
+ float sampL = 0.0;
+ float sampR = 0.0;
+ for (uint32_t j = 0; j < channels; j++) {
+ sampL += in[i*mIn.Channels()+j]*dmatrix[mIn.Channels()-3][j][0];
+ sampR += in[i*mIn.Channels()+j]*dmatrix[mIn.Channels()-3][j][1];
+ }
+ *out++ = sampL;
+ *out++ = sampR;
+ }
+ } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
+ // Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
+ // Coefficients in Q14.
+ static const int16_t dmatrix[6][8][2]= {
+ /*3*/{{9598, 0},{0, 9598},{6786,6786}},
+ /*4*/{{6925, 0},{0, 6925},{5997,3462},{3462,5997}},
+ /*5*/{{10663,0},{0, 10663},{7540,7540},{9234,5331},{5331,9234}},
+ /*6*/{{8668, 0},{0, 8668},{6129,6129},{6129,6129},{7507,4335},{4335,7507}},
+ /*7*/{{7459, 0},{0, 7459},{5275,5275},{5275,5275},{4568,4568},{6460,3731},{3731,6460}},
+ /*8*/{{6368, 0},{0, 6368},{4502,4502},{4502,4502},{5514,3184},{3184,5514},{5514,3184},{3184,5514}}
+ };
+ // Re-write the buffer with downmixed data
+ const int16_t* in = static_cast<const int16_t*>(aIn);
+ int16_t* out = static_cast<int16_t*>(aOut);
+ for (uint32_t i = 0; i < aFrames; i++) {
+ int32_t sampL = 0;
+ int32_t sampR = 0;
+ for (uint32_t j = 0; j < channels; j++) {
+ sampL+=in[i*channels+j]*dmatrix[channels-3][j][0];
+ sampR+=in[i*channels+j]*dmatrix[channels-3][j][1];
+ }
+ *out++ = clipTo15((sampL + 8192)>>14);
+ *out++ = clipTo15((sampR + 8192)>>14);
+ }
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ }
+
+ // If we are to continue downmixing to mono, start working on the output
+ // buffer.
+ aIn = aOut;
+ channels = 2;
+ }
+
+ if (mOut.Channels() == 1) {
+ if (mIn.Format() == AudioConfig::FORMAT_FLT) {
+ const float* in = static_cast<const float*>(aIn);
+ float* out = static_cast<float*>(aOut);
+ for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
+ float sample = 0.0;
+ // The sample of the buffer would be interleaved.
+ sample = (in[fIdx*channels] + in[fIdx*channels + 1]) * 0.5;
+ *out++ = sample;
+ }
+ } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
+ const int16_t* in = static_cast<const int16_t*>(aIn);
+ int16_t* out = static_cast<int16_t*>(aOut);
+ for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
+ int32_t sample = 0.0;
+ // The sample of the buffer would be interleaved.
+ sample = (in[fIdx*channels] + in[fIdx*channels + 1]) * 0.5;
+ *out++ = sample;
+ }
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ }
+ }
+ return aFrames;
+}
+
+size_t
+AudioConverter::ResampleAudio(void* aOut, const void* aIn, size_t aFrames)
+{
+ if (!mResampler) {
+ return 0;
+ }
+ uint32_t outframes = ResampleRecipientFrames(aFrames);
+ uint32_t inframes = aFrames;
+
+ int error;
+ if (mOut.Format() == AudioConfig::FORMAT_FLT) {
+ const float* in = reinterpret_cast<const float*>(aIn);
+ float* out = reinterpret_cast<float*>(aOut);
+ error =
+ speex_resampler_process_interleaved_float(mResampler, in, &inframes,
+ out, &outframes);
+ } else if (mOut.Format() == AudioConfig::FORMAT_S16) {
+ const int16_t* in = reinterpret_cast<const int16_t*>(aIn);
+ int16_t* out = reinterpret_cast<int16_t*>(aOut);
+ error =
+ speex_resampler_process_interleaved_int(mResampler, in, &inframes,
+ out, &outframes);
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ error = RESAMPLER_ERR_ALLOC_FAILED;
+ }
+ MOZ_ASSERT(error == RESAMPLER_ERR_SUCCESS);
+ if (error != RESAMPLER_ERR_SUCCESS) {
+ speex_resampler_destroy(mResampler);
+ mResampler = nullptr;
+ return 0;
+ }
+ MOZ_ASSERT(inframes == aFrames, "Some frames will be dropped");
+ return outframes;
+}
+
+void
+AudioConverter::RecreateResampler()
+{
+ if (mResampler) {
+ speex_resampler_destroy(mResampler);
+ }
+ int error;
+ mResampler = speex_resampler_init(mOut.Channels(),
+ mIn.Rate(),
+ mOut.Rate(),
+ SPEEX_RESAMPLER_QUALITY_DEFAULT,
+ &error);
+
+ if (error == RESAMPLER_ERR_SUCCESS) {
+ speex_resampler_skip_zeros(mResampler);
+ } else {
+ NS_WARNING("Failed to initialize resampler.");
+ mResampler = nullptr;
+ }
+}
+
+size_t
+AudioConverter::DrainResampler(void* aOut)
+{
+ if (!mResampler) {
+ return 0;
+ }
+ int frames = speex_resampler_get_input_latency(mResampler);
+ AlignedByteBuffer buffer(FramesOutToBytes(frames));
+ if (!buffer) {
+ // OOM
+ return 0;
+ }
+ frames = ResampleAudio(aOut, buffer.Data(), frames);
+ // Tore down the resampler as it's easier than handling follow-up.
+ RecreateResampler();
+ return frames;
+}
+
+size_t
+AudioConverter::UpmixAudio(void* aOut, const void* aIn, size_t aFrames) const
+{
+ MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
+ mIn.Format() == AudioConfig::FORMAT_FLT);
+ MOZ_ASSERT(mIn.Channels() < mOut.Channels());
+ MOZ_ASSERT(mIn.Channels() == 1, "Can only upmix mono for now");
+ MOZ_ASSERT(mOut.Channels() == 2, "Can only upmix to stereo for now");
+
+ if (mOut.Channels() != 2) {
+ return 0;
+ }
+
+ // Upmix mono to stereo.
+ // This is a very dumb mono to stereo upmixing, power levels are preserved
+ // following the calculation: left = right = -3dB*mono.
+ if (mIn.Format() == AudioConfig::FORMAT_FLT) {
+ const float m3db = std::sqrt(0.5); // -3dB = sqrt(1/2)
+ const float* in = static_cast<const float*>(aIn);
+ float* out = static_cast<float*>(aOut);
+ for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
+ float sample = in[fIdx] * m3db;
+ // The samples of the buffer would be interleaved.
+ *out++ = sample;
+ *out++ = sample;
+ }
+ } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
+ const int16_t* in = static_cast<const int16_t*>(aIn);
+ int16_t* out = static_cast<int16_t*>(aOut);
+ for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
+ int16_t sample = ((int32_t)in[fIdx] * 11585) >> 14; // close enough to i*sqrt(0.5)
+ // The samples of the buffer would be interleaved.
+ *out++ = sample;
+ *out++ = sample;
+ }
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ }
+
+ return aFrames;
+}
+
+size_t
+AudioConverter::ResampleRecipientFrames(size_t aFrames) const
+{
+ if (!aFrames && mIn.Rate() != mOut.Rate()) {
+ // The resampler will be drained, account for frames currently buffered
+ // in the resampler.
+ if (!mResampler) {
+ return 0;
+ }
+ return speex_resampler_get_output_latency(mResampler);
+ } else {
+ return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;
+ }
+}
+
+size_t
+AudioConverter::FramesOutToSamples(size_t aFrames) const
+{
+ return aFrames * mOut.Channels();
+}
+
+size_t
+AudioConverter::SamplesInToFrames(size_t aSamples) const
+{
+ return aSamples / mIn.Channels();
+}
+
+size_t
+AudioConverter::FramesOutToBytes(size_t aFrames) const
+{
+ return FramesOutToSamples(aFrames) * AudioConfig::SampleSize(mOut.Format());
+}
+} // namespace mozilla