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abe07cabea
Enable WebRTC unit tests to be built using standalone WebRTC library Includes VideoSegment and SimpleImageBuffer. --HG-- extra : rebase_source : 8eeea15e41fd298baef1ef1a2bc3f60182962861
218 lines
7.8 KiB
C++
218 lines
7.8 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AudioSegment.h"
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#include "AudioStream.h"
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#include "AudioMixer.h"
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#include "AudioChannelFormat.h"
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#include "Latency.h"
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#include <speex/speex_resampler.h>
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namespace mozilla {
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template <class SrcT, class DestT>
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static void
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InterleaveAndConvertBuffer(const SrcT** aSourceChannels,
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int32_t aLength, float aVolume,
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int32_t aChannels,
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DestT* aOutput)
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{
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DestT* output = aOutput;
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for (int32_t i = 0; i < aLength; ++i) {
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for (int32_t channel = 0; channel < aChannels; ++channel) {
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float v = AudioSampleToFloat(aSourceChannels[channel][i])*aVolume;
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*output = FloatToAudioSample<DestT>(v);
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++output;
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}
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}
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}
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void
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InterleaveAndConvertBuffer(const void** aSourceChannels,
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AudioSampleFormat aSourceFormat,
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int32_t aLength, float aVolume,
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int32_t aChannels,
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AudioDataValue* aOutput)
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{
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switch (aSourceFormat) {
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case AUDIO_FORMAT_FLOAT32:
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InterleaveAndConvertBuffer(reinterpret_cast<const float**>(aSourceChannels),
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aLength,
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aVolume,
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aChannels,
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aOutput);
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break;
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case AUDIO_FORMAT_S16:
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InterleaveAndConvertBuffer(reinterpret_cast<const int16_t**>(aSourceChannels),
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aLength,
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aVolume,
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aChannels,
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aOutput);
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break;
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case AUDIO_FORMAT_SILENCE:
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// nothing to do here.
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break;
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}
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}
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void
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AudioSegment::ApplyVolume(float aVolume)
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{
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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ci->mVolume *= aVolume;
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}
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}
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static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
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static const uint8_t gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*AUDIO_PROCESSING_FRAMES] = {0};
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void
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DownmixAndInterleave(const nsTArray<const void*>& aChannelData,
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AudioSampleFormat aSourceFormat, int32_t aDuration,
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float aVolume, uint32_t aOutputChannels,
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AudioDataValue* aOutput)
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{
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nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;
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nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixConversionBuffer;
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nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixOutputBuffer;
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channelData.SetLength(aChannelData.Length());
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if (aSourceFormat != AUDIO_FORMAT_FLOAT32) {
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NS_ASSERTION(aSourceFormat == AUDIO_FORMAT_S16, "unknown format");
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downmixConversionBuffer.SetLength(aDuration*aChannelData.Length());
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for (uint32_t i = 0; i < aChannelData.Length(); ++i) {
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float* conversionBuf = downmixConversionBuffer.Elements() + (i*aDuration);
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const int16_t* sourceBuf = static_cast<const int16_t*>(aChannelData[i]);
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for (uint32_t j = 0; j < (uint32_t)aDuration; ++j) {
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conversionBuf[j] = AudioSampleToFloat(sourceBuf[j]);
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}
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channelData[i] = conversionBuf;
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}
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} else {
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for (uint32_t i = 0; i < aChannelData.Length(); ++i) {
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channelData[i] = aChannelData[i];
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}
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}
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downmixOutputBuffer.SetLength(aDuration*aOutputChannels);
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nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannelBuffers;
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nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> outputChannelData;
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outputChannelBuffers.SetLength(aOutputChannels);
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outputChannelData.SetLength(aOutputChannels);
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for (uint32_t i = 0; i < (uint32_t)aOutputChannels; ++i) {
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outputChannelData[i] = outputChannelBuffers[i] =
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downmixOutputBuffer.Elements() + aDuration*i;
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}
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if (channelData.Length() > aOutputChannels) {
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AudioChannelsDownMix(channelData, outputChannelBuffers.Elements(),
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aOutputChannels, aDuration);
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}
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InterleaveAndConvertBuffer(outputChannelData.Elements(), AUDIO_FORMAT_FLOAT32,
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aDuration, aVolume, aOutputChannels, aOutput);
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}
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void AudioSegment::ResampleChunks(SpeexResamplerState* aResampler, uint32_t aInRate, uint32_t aOutRate)
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{
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if (mChunks.IsEmpty()) {
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return;
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}
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MOZ_ASSERT(aResampler || IsNull(), "We can only be here without a resampler if this segment is null.");
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AudioSampleFormat format = AUDIO_FORMAT_SILENCE;
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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if (ci->mBufferFormat != AUDIO_FORMAT_SILENCE) {
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format = ci->mBufferFormat;
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}
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}
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switch (format) {
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// If the format is silence at this point, all the chunks are silent. The
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// actual function we use does not matter, it's just a matter of changing
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// the chunks duration.
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case AUDIO_FORMAT_SILENCE:
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case AUDIO_FORMAT_FLOAT32:
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Resample<float>(aResampler, aInRate, aOutRate);
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break;
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case AUDIO_FORMAT_S16:
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Resample<int16_t>(aResampler, aInRate, aOutRate);
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break;
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default:
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MOZ_ASSERT(false);
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break;
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}
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}
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void
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AudioSegment::WriteTo(uint64_t aID, AudioMixer& aMixer, uint32_t aOutputChannels, uint32_t aSampleRate)
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{
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nsAutoTArray<AudioDataValue,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> buf;
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nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;
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// Offset in the buffer that will end up sent to the AudioStream, in samples.
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uint32_t offset = 0;
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if (GetDuration() <= 0) {
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MOZ_ASSERT(GetDuration() == 0);
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return;
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}
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uint32_t outBufferLength = GetDuration() * aOutputChannels;
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buf.SetLength(outBufferLength);
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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AudioChunk& c = *ci;
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uint32_t frames = c.mDuration;
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// If we have written data in the past, or we have real (non-silent) data
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// to write, we can proceed. Otherwise, it means we just started the
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// AudioStream, and we don't have real data to write to it (just silence).
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// To avoid overbuffering in the AudioStream, we simply drop the silence,
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// here. The stream will underrun and output silence anyways.
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if (c.mBuffer && c.mBufferFormat != AUDIO_FORMAT_SILENCE) {
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channelData.SetLength(c.mChannelData.Length());
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for (uint32_t i = 0; i < channelData.Length(); ++i) {
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channelData[i] = c.mChannelData[i];
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}
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if (channelData.Length() < aOutputChannels) {
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// Up-mix. Note that this might actually make channelData have more
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// than aOutputChannels temporarily.
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AudioChannelsUpMix(&channelData, aOutputChannels, gZeroChannel);
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}
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if (channelData.Length() > aOutputChannels) {
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// Down-mix.
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DownmixAndInterleave(channelData, c.mBufferFormat, frames,
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c.mVolume, aOutputChannels, buf.Elements() + offset);
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} else {
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InterleaveAndConvertBuffer(channelData.Elements(), c.mBufferFormat,
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frames, c.mVolume,
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aOutputChannels,
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buf.Elements() + offset);
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}
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} else {
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// Assumes that a bit pattern of zeroes == 0.0f
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memset(buf.Elements() + offset, 0, aOutputChannels * frames * sizeof(AudioDataValue));
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}
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offset += frames * aOutputChannels;
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#if !defined(MOZILLA_XPCOMRT_API)
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if (!c.mTimeStamp.IsNull()) {
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TimeStamp now = TimeStamp::Now();
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// would be more efficient to c.mTimeStamp to ms on create time then pass here
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LogTime(AsyncLatencyLogger::AudioMediaStreamTrack, aID,
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(now - c.mTimeStamp).ToMilliseconds(), c.mTimeStamp);
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}
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#endif // !defined(MOZILLA_XPCOMRT_API)
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}
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if (offset) {
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aMixer.Mix(buf.Elements(), aOutputChannels, offset / aOutputChannels, aSampleRate);
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}
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}
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}
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