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-rw-r--r--dom/media/AudioConverter.cpp480
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diff --git a/dom/media/AudioConverter.cpp b/dom/media/AudioConverter.cpp
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+++ 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 <speex/speex_resampler.h>
+#include <string.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(CanConvert(aIn, aOut),
+ "The conversion is 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::CanConvert(const AudioConfig& aIn,
+ const AudioConfig& aOut) {
+ if (aIn.Format() != aOut.Format() ||
+ aIn.Interleaved() != aOut.Interleaved()) {
+ NS_WARNING("No format conversion is supported at this stage");
+ return false;
+ }
+ if (aIn.Channels() != aOut.Channels() && aOut.Channels() > 2) {
+ NS_WARNING(
+ "Only down/upmixing to mono or stereo is supported at this stage");
+ return false;
+ }
+ if (!aOut.Interleaved()) {
+ NS_WARNING("planar audio format not supported");
+ return false;
+ }
+ return true;
+}
+
+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 (!aFrames) {
+ return 0;
+ }
+ 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 <= AudioConfig::ChannelLayout::MAX_CHANNELS);
+ AudioDataType val[AudioConfig::ChannelLayout::MAX_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());
+ MOZ_DIAGNOSTIC_ASSERT(CanReorderAudio());
+
+ if (mChannelOrderMap.IsEmpty() || mOut.Channels() == 1 ||
+ mOut.Layout() == mIn.Layout()) {
+ // If channel count is 1, planar and non-planar formats are the same or
+ // there's nothing to reorder, or if we don't know how to re-order.
+ 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.Elements());
+ break;
+ case 16:
+ _ReOrderInterleavedChannels((int16_t*)aOut, (const int16_t*)aIn, aFrames,
+ mIn.Channels(), mChannelOrderMap.Elements());
+ break;
+ default:
+ MOZ_DIAGNOSTIC_ASSERT(AudioConfig::SampleSize(mOut.Format()) == 4);
+ _ReOrderInterleavedChannels((int32_t*)aOut, (const int32_t*)aIn, aFrames,
+ mIn.Channels(), mChannelOrderMap.Elements());
+ break;
+ }
+}
+
+static inline int16_t clipTo15(int32_t aX) {
+ return aX < -32768 ? -32768 : aX <= 32767 ? aX : 32767;
+}
+
+template <typename TYPE>
+static void dumbUpDownMix(TYPE* aOut, int32_t aOutChannels, const TYPE* aIn,
+ int32_t aInChannels, int32_t aFrames) {
+ if (aIn == aOut) {
+ return;
+ }
+ int32_t commonChannels = std::min(aInChannels, aOutChannels);
+
+ for (int32_t i = 0; i < aFrames; i++) {
+ for (int32_t j = 0; j < commonChannels; j++) {
+ aOut[i * aOutChannels + j] = aIn[i * aInChannels + j];
+ }
+ if (aOutChannels > aInChannels) {
+ for (int32_t j = 0; j < aInChannels - aOutChannels; j++) {
+ aOut[i * aOutChannels + j] = 0;
+ }
+ }
+ }
+}
+
+size_t AudioConverter::DownmixAudio(void* aOut, const void* aIn,
+ size_t aFrames) const {
+ MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
+ mIn.Format() == AudioConfig::FORMAT_FLT);
+ MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() >= mOut.Channels());
+ MOZ_DIAGNOSTIC_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
+ mOut.Layout() == AudioConfig::ChannelLayout(1));
+
+ uint32_t inChannels = mIn.Channels();
+ uint32_t outChannels = mOut.Channels();
+
+ if (inChannels == outChannels) {
+ if (aOut != aIn) {
+ memmove(aOut, aIn, FramesOutToBytes(aFrames));
+ }
+ return aFrames;
+ }
+
+ if (!mIn.Layout().IsValid() || !mOut.Layout().IsValid()) {
+ // Dumb copy dropping extra channels.
+ if (mIn.Format() == AudioConfig::FORMAT_FLT) {
+ dumbUpDownMix(static_cast<float*>(aOut), outChannels,
+ static_cast<const float*>(aIn), inChannels, aFrames);
+ } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
+ dumbUpDownMix(static_cast<int16_t*>(aOut), outChannels,
+ static_cast<const int16_t*>(aIn), inChannels, aFrames);
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ }
+ return aFrames;
+ }
+
+ MOZ_ASSERT(
+ mIn.Layout() == AudioConfig::ChannelLayout::SMPTEDefault(mIn.Layout()),
+ "Can only downmix input data in SMPTE layout");
+ if (inChannels > 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 < inChannels; j++) {
+ sampL += in[i * inChannels + j] * dmatrix[inChannels - 3][j][0];
+ sampR += in[i * inChannels + j] * dmatrix[inChannels - 3][j][1];
+ }
+ if (outChannels == 2) {
+ *out++ = sampL;
+ *out++ = sampR;
+ } else {
+ *out++ = (sampL + sampR) * 0.5;
+ }
+ }
+ } 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 < inChannels; j++) {
+ sampL += in[i * inChannels + j] * dmatrix[inChannels - 3][j][0];
+ sampR += in[i * inChannels + j] * dmatrix[inChannels - 3][j][1];
+ }
+ sampL = clipTo15((sampL + 8192) >> 14);
+ sampR = clipTo15((sampR + 8192) >> 14);
+ if (outChannels == 2) {
+ *out++ = sampL;
+ *out++ = sampR;
+ } else {
+ *out++ = (sampL + sampR) * 0.5;
+ }
+ }
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ }
+ return aFrames;
+ }
+
+ MOZ_DIAGNOSTIC_ASSERT(inChannels == 2 && outChannels == 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 * inChannels] + in[fIdx * inChannels + 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 * inChannels] + in[fIdx * inChannels + 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 (!mIn.Layout().IsValid() || !mOut.Layout().IsValid() ||
+ mOut.Channels() != 2) {
+ // Dumb copy the channels and insert silence for the extra channels.
+ if (mIn.Format() == AudioConfig::FORMAT_FLT) {
+ dumbUpDownMix(static_cast<float*>(aOut), mOut.Channels(),
+ static_cast<const float*>(aIn), mIn.Channels(), aFrames);
+ } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
+ dumbUpDownMix(static_cast<int16_t*>(aOut), mOut.Channels(),
+ static_cast<const int16_t*>(aIn), mIn.Channels(), aFrames);
+ } else {
+ MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
+ }
+ return aFrames;
+ }
+
+ // 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()) {
+ if (!mResampler) {
+ return 0;
+ }
+ // We drain by pushing in get_input_latency() samples of 0
+ aFrames = speex_resampler_get_input_latency(mResampler);
+ }
+ 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
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