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Diffstat (limited to 'dom/media/AudioConverter.cpp')
| -rw-r--r-- | dom/media/AudioConverter.cpp | 393 |
1 files changed, 393 insertions, 0 deletions
diff --git a/dom/media/AudioConverter.cpp b/dom/media/AudioConverter.cpp new file mode 100644 index 000000000..77ad46ec6 --- /dev/null +++ b/dom/media/AudioConverter.cpp @@ -0,0 +1,393 @@ +/* -*- 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 |