/* -*- 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()) { 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