bug 1391482 generalize WebAudioDecodeJob buffer as AudioChunk r=padenot

MozReview-Commit-ID: 4H3F0SzAknc

--HG--
extra : rebase_source : 0c307d759617ee3bc2885d7e8683c36c650d2d4f
This commit is contained in:
Karl Tomlinson 2017-08-16 18:10:06 +12:00
Родитель 2697172632
Коммит 5de6db32a5
2 изменённых файлов: 29 добавлений и 25 удалений

Просмотреть файл

@ -340,14 +340,23 @@ MediaDecodeTask::FinishDecode()
// Allocate the channel buffers. Note that if we end up resampling, we may
// write fewer bytes than mResampledFrames to the output buffer, in which
// case mWriteIndex will tell us how many valid samples we have.
mDecodeJob.mBuffer = ThreadSharedFloatArrayBufferList::
// case writeIndex will tell us how many valid samples we have.
RefPtr<ThreadSharedFloatArrayBufferList> buffer =
ThreadSharedFloatArrayBufferList::
Create(channelCount, resampledFrames, fallible);
if (!mDecodeJob.mBuffer) {
if (!buffer) {
ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
return;
}
mDecodeJob.mBuffer.mChannelData.SetLength(channelCount);
for (uint32_t i = 0; i < channelCount; ++i) {
mDecodeJob.mBuffer.mChannelData[i] = buffer->GetData(i);
}
mDecodeJob.mBuffer.mBuffer = buffer.forget();
mDecodeJob.mBuffer.mVolume = 1.0f;
mDecodeJob.mBuffer.mBufferFormat = AUDIO_FORMAT_FLOAT32;
uint32_t writeIndex = 0;
RefPtr<AudioData> audioData;
while ((audioData = mAudioQueue.PopFront())) {
audioData->EnsureAudioBuffer(); // could lead to a copy :(
@ -355,33 +364,33 @@ MediaDecodeTask::FinishDecode()
(audioData->mAudioBuffer->Data());
if (sampleRate != destSampleRate) {
const uint32_t maxOutSamples = resampledFrames - mDecodeJob.mWriteIndex;
const uint32_t maxOutSamples = resampledFrames - writeIndex;
for (uint32_t i = 0; i < audioData->mChannels; ++i) {
uint32_t inSamples = audioData->mFrames;
uint32_t outSamples = maxOutSamples;
float* outData =
mDecodeJob.mBuffer->GetDataForWrite(i) + mDecodeJob.mWriteIndex;
mDecodeJob.mBuffer.ChannelDataForWrite<float>(i) + writeIndex;
WebAudioUtils::SpeexResamplerProcess(
resampler, i, &bufferData[i * audioData->mFrames], &inSamples,
outData, &outSamples);
if (i == audioData->mChannels - 1) {
mDecodeJob.mWriteIndex += outSamples;
MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
writeIndex += outSamples;
MOZ_ASSERT(writeIndex <= resampledFrames);
MOZ_ASSERT(inSamples == audioData->mFrames);
}
}
} else {
for (uint32_t i = 0; i < audioData->mChannels; ++i) {
float* outData =
mDecodeJob.mBuffer->GetDataForWrite(i) + mDecodeJob.mWriteIndex;
mDecodeJob.mBuffer.ChannelDataForWrite<float>(i) + writeIndex;
ConvertAudioSamples(&bufferData[i * audioData->mFrames],
outData, audioData->mFrames);
if (i == audioData->mChannels - 1) {
mDecodeJob.mWriteIndex += audioData->mFrames;
writeIndex += audioData->mFrames;
}
}
}
@ -389,25 +398,26 @@ MediaDecodeTask::FinishDecode()
if (sampleRate != destSampleRate) {
uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
const uint32_t maxOutSamples = resampledFrames - mDecodeJob.mWriteIndex;
const uint32_t maxOutSamples = resampledFrames - writeIndex;
for (uint32_t i = 0; i < channelCount; ++i) {
uint32_t inSamples = inputLatency;
uint32_t outSamples = maxOutSamples;
float* outData =
mDecodeJob.mBuffer->GetDataForWrite(i) + mDecodeJob.mWriteIndex;
mDecodeJob.mBuffer.ChannelDataForWrite<float>(i) + writeIndex;
WebAudioUtils::SpeexResamplerProcess(
resampler, i, (AudioDataValue*)nullptr, &inSamples,
outData, &outSamples);
if (i == channelCount - 1) {
mDecodeJob.mWriteIndex += outSamples;
MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
writeIndex += outSamples;
MOZ_ASSERT(writeIndex <= resampledFrames);
MOZ_ASSERT(inSamples == inputLatency);
}
}
}
mDecodeJob.mBuffer.mDuration = writeIndex;
mPhase = PhaseEnum::AllocateBuffer;
mMainThread->Dispatch(do_AddRef(this));
}
@ -444,12 +454,9 @@ WebAudioDecodeJob::AllocateBuffer()
MOZ_ASSERT(NS_IsMainThread());
// Now create the AudioBuffer
ErrorResult rv;
uint32_t channelCount = mBuffer->GetChannels();
mOutput = AudioBuffer::Create(mContext->GetOwner(), channelCount,
mWriteIndex, mContext->SampleRate(),
mBuffer.forget(), rv);
return !rv.Failed();
mOutput = AudioBuffer::Create(mContext->GetOwner(),
mContext->SampleRate(), Move(mBuffer));
return mOutput != nullptr;
}
void
@ -495,7 +502,6 @@ WebAudioDecodeJob::WebAudioDecodeJob(const nsACString& aContentType,
DecodeSuccessCallback* aSuccessCallback,
DecodeErrorCallback* aFailureCallback)
: mContentType(aContentType)
, mWriteIndex(0)
, mContext(aContext)
, mPromise(aPromise)
, mSuccessCallback(aSuccessCallback)
@ -595,9 +601,7 @@ WebAudioDecodeJob::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
if (mOutput) {
amount += mOutput->SizeOfIncludingThis(aMallocSizeOf);
}
if (mBuffer) {
amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
}
amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
return amount;
}

Просмотреть файл

@ -7,6 +7,7 @@
#ifndef MediaBufferDecoder_h_
#define MediaBufferDecoder_h_
#include "AudioSegment.h"
#include "nsWrapperCache.h"
#include "nsCOMPtr.h"
#include "nsString.h"
@ -55,14 +56,13 @@ struct WebAudioDecodeJob final
size_t SizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
AudioChunk mBuffer;
nsCString mContentType;
uint32_t mWriteIndex;
RefPtr<dom::AudioContext> mContext;
RefPtr<dom::Promise> mPromise;
RefPtr<dom::DecodeSuccessCallback> mSuccessCallback;
RefPtr<dom::DecodeErrorCallback> mFailureCallback; // can be null
RefPtr<dom::AudioBuffer> mOutput;
RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
};
void AsyncDecodeWebAudio(const char* aContentType, uint8_t* aBuffer,