зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1156472 - Part 13 - Make necessary adjustments for integer audio. r=jesup
This commit is contained in:
Родитель
67ca74db0b
Коммит
90a222f71a
|
@ -0,0 +1,133 @@
|
|||
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
|
||||
/* 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 "MediaStreamGraphImpl.h"
|
||||
#include "mozilla/MathAlgorithms.h"
|
||||
#include "mozilla/unused.h"
|
||||
|
||||
#include "AudioSegment.h"
|
||||
#include "mozilla/Logging.h"
|
||||
#include "mozilla/Attributes.h"
|
||||
#include "AudioCaptureStream.h"
|
||||
#include "ImageContainer.h"
|
||||
#include "AudioNodeEngine.h"
|
||||
#include "AudioNodeStream.h"
|
||||
#include "AudioNodeExternalInputStream.h"
|
||||
#include "webaudio/MediaStreamAudioDestinationNode.h"
|
||||
#include <algorithm>
|
||||
#include "DOMMediaStream.h"
|
||||
|
||||
using namespace mozilla::layers;
|
||||
using namespace mozilla::dom;
|
||||
using namespace mozilla::gfx;
|
||||
|
||||
namespace mozilla
|
||||
{
|
||||
|
||||
// We are mixing to mono until PeerConnection can accept stereo
|
||||
static const uint32_t MONO = 1;
|
||||
|
||||
AudioCaptureStream::AudioCaptureStream(DOMMediaStream* aWrapper)
|
||||
: ProcessedMediaStream(aWrapper), mTrackCreated(false)
|
||||
{
|
||||
MOZ_ASSERT(NS_IsMainThread());
|
||||
MOZ_COUNT_CTOR(AudioCaptureStream);
|
||||
mMixer.AddCallback(this);
|
||||
}
|
||||
|
||||
AudioCaptureStream::~AudioCaptureStream()
|
||||
{
|
||||
MOZ_COUNT_DTOR(AudioCaptureStream);
|
||||
mMixer.RemoveCallback(this);
|
||||
}
|
||||
|
||||
void
|
||||
AudioCaptureStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
|
||||
uint32_t aFlags)
|
||||
{
|
||||
uint32_t inputCount = mInputs.Length();
|
||||
StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK);
|
||||
// Notify the DOM everything is in order.
|
||||
if (!mTrackCreated) {
|
||||
for (uint32_t i = 0; i < mListeners.Length(); i++) {
|
||||
MediaStreamListener* l = mListeners[i];
|
||||
AudioSegment tmp;
|
||||
l->NotifyQueuedTrackChanges(
|
||||
Graph(), AUDIO_TRACK, 0, MediaStreamListener::TRACK_EVENT_CREATED, tmp);
|
||||
l->NotifyFinishedTrackCreation(Graph());
|
||||
}
|
||||
mTrackCreated = true;
|
||||
}
|
||||
|
||||
// If the captured stream is connected back to a object on the page (be it an
|
||||
// HTMLMediaElement with a stream as source, or an AudioContext), a cycle
|
||||
// situation occur. This can work if it's an AudioContext with at least one
|
||||
// DelayNode, but the MSG will mute the whole cycle otherwise.
|
||||
bool blocked = mFinished || mBlocked.GetAt(aFrom);
|
||||
if (blocked || InMutedCycle() || inputCount == 0) {
|
||||
track->Get<AudioSegment>()->AppendNullData(aTo - aFrom);
|
||||
} else {
|
||||
// We mix down all the tracks of all inputs, to a stereo track. Everything
|
||||
// is {up,down}-mixed to stereo.
|
||||
mMixer.StartMixing();
|
||||
AudioSegment output;
|
||||
for (uint32_t i = 0; i < inputCount; i++) {
|
||||
MediaStream* s = mInputs[i]->GetSource();
|
||||
StreamBuffer::TrackIter tracks(s->GetStreamBuffer(), MediaSegment::AUDIO);
|
||||
while (!tracks.IsEnded()) {
|
||||
AudioSegment* inputSegment = tracks->Get<AudioSegment>();
|
||||
StreamTime inputStart = s->GraphTimeToStreamTime(aFrom);
|
||||
StreamTime inputEnd = s->GraphTimeToStreamTime(aTo);
|
||||
AudioSegment toMix;
|
||||
toMix.AppendSlice(*inputSegment, inputStart, inputEnd);
|
||||
// Care for streams blocked in the [aTo, aFrom] range.
|
||||
if (inputEnd - inputStart < aTo - aFrom) {
|
||||
toMix.AppendNullData((aTo - aFrom) - (inputEnd - inputStart));
|
||||
}
|
||||
toMix.Mix(mMixer, MONO, Graph()->GraphRate());
|
||||
tracks.Next();
|
||||
}
|
||||
}
|
||||
// This calls MixerCallback below
|
||||
mMixer.FinishMixing();
|
||||
}
|
||||
|
||||
// Regardless of the status of the input tracks, we go foward.
|
||||
mBuffer.AdvanceKnownTracksTime(GraphTimeToStreamTime((aTo)));
|
||||
}
|
||||
|
||||
void
|
||||
AudioCaptureStream::MixerCallback(AudioDataValue* aMixedBuffer,
|
||||
AudioSampleFormat aFormat, uint32_t aChannels,
|
||||
uint32_t aFrames, uint32_t aSampleRate)
|
||||
{
|
||||
nsAutoTArray<nsTArray<AudioDataValue>, MONO> output;
|
||||
nsAutoTArray<const AudioDataValue*, MONO> bufferPtrs;
|
||||
output.SetLength(MONO);
|
||||
bufferPtrs.SetLength(MONO);
|
||||
|
||||
uint32_t written = 0;
|
||||
// We need to copy here, because the mixer will reuse the storage, we should
|
||||
// not hold onto it. Buffers are in planar format.
|
||||
for (uint32_t channel = 0; channel < aChannels; channel++) {
|
||||
AudioDataValue* out = output[channel].AppendElements(aFrames);
|
||||
PodCopy(out, aMixedBuffer + written, aFrames);
|
||||
bufferPtrs[channel] = out;
|
||||
written += aFrames;
|
||||
}
|
||||
AudioChunk chunk;
|
||||
chunk.mBuffer = new mozilla::SharedChannelArrayBuffer<AudioDataValue>(&output);
|
||||
chunk.mDuration = aFrames;
|
||||
chunk.mBufferFormat = aFormat;
|
||||
chunk.mVolume = 1.0f;
|
||||
chunk.mChannelData.SetLength(MONO);
|
||||
for (uint32_t channel = 0; channel < aChannels; channel++) {
|
||||
chunk.mChannelData[channel] = bufferPtrs[channel];
|
||||
}
|
||||
|
||||
// Now we have mixed data, simply append it to out track.
|
||||
EnsureTrack(AUDIO_TRACK)->Get<AudioSegment>()->AppendAndConsumeChunk(&chunk);
|
||||
}
|
||||
}
|
|
@ -4,26 +4,11 @@
|
|||
* You can obtain one at http://mozilla.org/MPL/2.0/. */
|
||||
|
||||
#include "AudioChannelFormat.h"
|
||||
#include "nsTArray.h"
|
||||
|
||||
#include <algorithm>
|
||||
|
||||
namespace mozilla {
|
||||
|
||||
enum {
|
||||
SURROUND_L,
|
||||
SURROUND_R,
|
||||
SURROUND_C,
|
||||
SURROUND_LFE,
|
||||
SURROUND_SL,
|
||||
SURROUND_SR
|
||||
};
|
||||
|
||||
static const uint32_t CUSTOM_CHANNEL_LAYOUTS = 6;
|
||||
|
||||
static const int IGNORE = CUSTOM_CHANNEL_LAYOUTS;
|
||||
static const float IGNORE_F = 0.0f;
|
||||
|
||||
uint32_t
|
||||
GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2)
|
||||
{
|
||||
|
@ -63,9 +48,6 @@ gUpMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
|
|||
{ { 0, 1, 2, 3, 4, IGNORE } }
|
||||
};
|
||||
|
||||
static const int gMixingMatrixIndexByChannels[CUSTOM_CHANNEL_LAYOUTS - 1] =
|
||||
{ 0, 5, 9, 12, 14 };
|
||||
|
||||
void
|
||||
AudioChannelsUpMix(nsTArray<const void*>* aChannelArray,
|
||||
uint32_t aOutputChannelCount,
|
||||
|
@ -108,94 +90,4 @@ AudioChannelsUpMix(nsTArray<const void*>* aChannelArray,
|
|||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* DownMixMatrix represents a conversion matrix efficiently by exploiting the
|
||||
* fact that each input channel contributes to at most one output channel,
|
||||
* except possibly for the C input channel in layouts that have one. Also,
|
||||
* every input channel is multiplied by the same coefficient for every output
|
||||
* channel it contributes to.
|
||||
*/
|
||||
struct DownMixMatrix {
|
||||
// Every input channel c is copied to output channel mInputDestination[c]
|
||||
// after multiplying by mInputCoefficient[c].
|
||||
uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
|
||||
// If not IGNORE, then the C channel is copied to this output channel after
|
||||
// multiplying by its coefficient.
|
||||
uint8_t mCExtraDestination;
|
||||
float mInputCoefficient[CUSTOM_CHANNEL_LAYOUTS];
|
||||
};
|
||||
|
||||
static const DownMixMatrix
|
||||
gDownMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
|
||||
{
|
||||
// Downmixes to mono
|
||||
{ { 0, 0 }, IGNORE, { 0.5f, 0.5f } },
|
||||
{ { 0, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 0, 0, 0 }, IGNORE, { 0.25f, 0.25f, 0.25f, 0.25f } },
|
||||
{ { 0, IGNORE, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 0, 0, IGNORE, 0, 0 }, IGNORE, { 0.7071f, 0.7071f, 1.0f, IGNORE_F, 0.5f, 0.5f } },
|
||||
// Downmixes to stereo
|
||||
{ { 0, 1, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F } },
|
||||
{ { 0, 1, 0, 1 }, IGNORE, { 0.5f, 0.5f, 0.5f, 0.5f } },
|
||||
{ { 0, 1, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 1, 0, IGNORE, 0, 1 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 0.7071f, 0.7071f } },
|
||||
// Downmixes to 3-channel
|
||||
{ { 0, 1, 2, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F } },
|
||||
{ { 0, 1, 2, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 1, 2, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
|
||||
// Downmixes to quad
|
||||
{ { 0, 1, 2, 3, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } },
|
||||
{ { 0, 1, 0, IGNORE, 2, 3 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 1.0f, 1.0f } },
|
||||
// Downmixes to 5-channel
|
||||
{ { 0, 1, 2, 3, 4, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } }
|
||||
};
|
||||
|
||||
void
|
||||
AudioChannelsDownMix(const nsTArray<const void*>& aChannelArray,
|
||||
float** aOutputChannels,
|
||||
uint32_t aOutputChannelCount,
|
||||
uint32_t aDuration)
|
||||
{
|
||||
uint32_t inputChannelCount = aChannelArray.Length();
|
||||
const void* const* inputChannels = aChannelArray.Elements();
|
||||
NS_ASSERTION(inputChannelCount > aOutputChannelCount, "Nothing to do");
|
||||
|
||||
if (inputChannelCount > 6) {
|
||||
// Just drop the unknown channels.
|
||||
for (uint32_t o = 0; o < aOutputChannelCount; ++o) {
|
||||
memcpy(aOutputChannels[o], inputChannels[o], aDuration*sizeof(float));
|
||||
}
|
||||
return;
|
||||
}
|
||||
|
||||
// Ignore unknown channels, they're just dropped.
|
||||
inputChannelCount = std::min<uint32_t>(6, inputChannelCount);
|
||||
|
||||
const DownMixMatrix& m = gDownMixMatrices[
|
||||
gMixingMatrixIndexByChannels[aOutputChannelCount - 1] +
|
||||
inputChannelCount - aOutputChannelCount - 1];
|
||||
|
||||
// This is slow, but general. We can define custom code for special
|
||||
// cases later.
|
||||
for (uint32_t s = 0; s < aDuration; ++s) {
|
||||
// Reserve an extra junk channel at the end for the cases where we
|
||||
// want an input channel to contribute to nothing
|
||||
float outputChannels[CUSTOM_CHANNEL_LAYOUTS + 1];
|
||||
memset(outputChannels, 0, sizeof(float)*(CUSTOM_CHANNEL_LAYOUTS));
|
||||
for (uint32_t c = 0; c < inputChannelCount; ++c) {
|
||||
outputChannels[m.mInputDestination[c]] +=
|
||||
m.mInputCoefficient[c]*(static_cast<const float*>(inputChannels[c]))[s];
|
||||
}
|
||||
// Utilize the fact that in every layout, C is the third channel.
|
||||
if (m.mCExtraDestination != IGNORE) {
|
||||
outputChannels[m.mCExtraDestination] +=
|
||||
m.mInputCoefficient[SURROUND_C]*(static_cast<const float*>(inputChannels[SURROUND_C]))[s];
|
||||
}
|
||||
|
||||
for (uint32_t c = 0; c < aOutputChannelCount; ++c) {
|
||||
aOutputChannels[c][s] = outputChannels[c];
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
} // namespace mozilla
|
||||
|
|
|
@ -9,6 +9,8 @@
|
|||
#include <stdint.h>
|
||||
|
||||
#include "nsTArrayForwardDeclare.h"
|
||||
#include "AudioSampleFormat.h"
|
||||
#include "nsTArray.h"
|
||||
|
||||
namespace mozilla {
|
||||
|
||||
|
@ -29,6 +31,26 @@ namespace mozilla {
|
|||
* Only 1, 2, 4 and 6 are currently defined in Web Audio.
|
||||
*/
|
||||
|
||||
enum {
|
||||
SURROUND_L,
|
||||
SURROUND_R,
|
||||
SURROUND_C,
|
||||
SURROUND_LFE,
|
||||
SURROUND_SL,
|
||||
SURROUND_SR
|
||||
};
|
||||
|
||||
const uint32_t CUSTOM_CHANNEL_LAYOUTS = 6;
|
||||
|
||||
// This is defined by some Windows SDK header.
|
||||
#undef IGNORE
|
||||
|
||||
const int IGNORE = CUSTOM_CHANNEL_LAYOUTS;
|
||||
const float IGNORE_F = 0.0f;
|
||||
|
||||
const int gMixingMatrixIndexByChannels[CUSTOM_CHANNEL_LAYOUTS - 1] =
|
||||
{ 0, 5, 9, 12, 14 };
|
||||
|
||||
/**
|
||||
* Return a channel count whose channel layout includes all the channels from
|
||||
* aChannels1 and aChannels2.
|
||||
|
@ -53,19 +75,102 @@ AudioChannelsUpMix(nsTArray<const void*>* aChannelArray,
|
|||
uint32_t aOutputChannelCount,
|
||||
const void* aZeroChannel);
|
||||
|
||||
/**
|
||||
* Given an array of input channels (which must be float format!),
|
||||
* downmix to aOutputChannelCount, and copy the results to the
|
||||
* channel buffers in aOutputChannels.
|
||||
* Don't call this with input count <= output count.
|
||||
*/
|
||||
void
|
||||
AudioChannelsDownMix(const nsTArray<const void*>& aChannelArray,
|
||||
float** aOutputChannels,
|
||||
uint32_t aOutputChannelCount,
|
||||
uint32_t aDuration);
|
||||
|
||||
// A version of AudioChannelsDownMix that downmixes int16_ts may be required.
|
||||
/**
|
||||
* DownMixMatrix represents a conversion matrix efficiently by exploiting the
|
||||
* fact that each input channel contributes to at most one output channel,
|
||||
* except possibly for the C input channel in layouts that have one. Also,
|
||||
* every input channel is multiplied by the same coefficient for every output
|
||||
* channel it contributes to.
|
||||
*/
|
||||
struct DownMixMatrix {
|
||||
// Every input channel c is copied to output channel mInputDestination[c]
|
||||
// after multiplying by mInputCoefficient[c].
|
||||
uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
|
||||
// If not IGNORE, then the C channel is copied to this output channel after
|
||||
// multiplying by its coefficient.
|
||||
uint8_t mCExtraDestination;
|
||||
float mInputCoefficient[CUSTOM_CHANNEL_LAYOUTS];
|
||||
};
|
||||
|
||||
static const DownMixMatrix
|
||||
gDownMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
|
||||
{
|
||||
// Downmixes to mono
|
||||
{ { 0, 0 }, IGNORE, { 0.5f, 0.5f } },
|
||||
{ { 0, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 0, 0, 0 }, IGNORE, { 0.25f, 0.25f, 0.25f, 0.25f } },
|
||||
{ { 0, IGNORE, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 0, 0, IGNORE, 0, 0 }, IGNORE, { 0.7071f, 0.7071f, 1.0f, IGNORE_F, 0.5f, 0.5f } },
|
||||
// Downmixes to stereo
|
||||
{ { 0, 1, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F } },
|
||||
{ { 0, 1, 0, 1 }, IGNORE, { 0.5f, 0.5f, 0.5f, 0.5f } },
|
||||
{ { 0, 1, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 1, 0, IGNORE, 0, 1 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 0.7071f, 0.7071f } },
|
||||
// Downmixes to 3-channel
|
||||
{ { 0, 1, 2, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F } },
|
||||
{ { 0, 1, 2, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F } },
|
||||
{ { 0, 1, 2, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
|
||||
// Downmixes to quad
|
||||
{ { 0, 1, 2, 3, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } },
|
||||
{ { 0, 1, 0, IGNORE, 2, 3 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 1.0f, 1.0f } },
|
||||
// Downmixes to 5-channel
|
||||
{ { 0, 1, 2, 3, 4, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } }
|
||||
};
|
||||
|
||||
/**
|
||||
* Given an array of input channels, downmix to aOutputChannelCount, and copy
|
||||
* the results to the channel buffers in aOutputChannels. Don't call this with
|
||||
* input count <= output count.
|
||||
*/
|
||||
template<typename T>
|
||||
void AudioChannelsDownMix(const nsTArray<const void*>& aChannelArray,
|
||||
T** aOutputChannels,
|
||||
uint32_t aOutputChannelCount,
|
||||
uint32_t aDuration)
|
||||
{
|
||||
uint32_t inputChannelCount = aChannelArray.Length();
|
||||
const void* const* inputChannels = aChannelArray.Elements();
|
||||
NS_ASSERTION(inputChannelCount > aOutputChannelCount, "Nothing to do");
|
||||
|
||||
if (inputChannelCount > 6) {
|
||||
// Just drop the unknown channels.
|
||||
for (uint32_t o = 0; o < aOutputChannelCount; ++o) {
|
||||
memcpy(aOutputChannels[o], inputChannels[o], aDuration*sizeof(T));
|
||||
}
|
||||
return;
|
||||
}
|
||||
|
||||
// Ignore unknown channels, they're just dropped.
|
||||
inputChannelCount = std::min<uint32_t>(6, inputChannelCount);
|
||||
|
||||
const DownMixMatrix& m = gDownMixMatrices[
|
||||
gMixingMatrixIndexByChannels[aOutputChannelCount - 1] +
|
||||
inputChannelCount - aOutputChannelCount - 1];
|
||||
|
||||
// This is slow, but general. We can define custom code for special
|
||||
// cases later.
|
||||
for (uint32_t s = 0; s < aDuration; ++s) {
|
||||
// Reserve an extra junk channel at the end for the cases where we
|
||||
// want an input channel to contribute to nothing
|
||||
T outputChannels[CUSTOM_CHANNEL_LAYOUTS + 1];
|
||||
memset(outputChannels, 0, sizeof(T)*(CUSTOM_CHANNEL_LAYOUTS));
|
||||
for (uint32_t c = 0; c < inputChannelCount; ++c) {
|
||||
outputChannels[m.mInputDestination[c]] +=
|
||||
m.mInputCoefficient[c]*(static_cast<const T*>(inputChannels[c]))[s];
|
||||
}
|
||||
// Utilize the fact that in every layout, C is the third channel.
|
||||
if (m.mCExtraDestination != IGNORE) {
|
||||
outputChannels[m.mCExtraDestination] +=
|
||||
m.mInputCoefficient[SURROUND_C]*(static_cast<const T*>(inputChannels[SURROUND_C]))[s];
|
||||
}
|
||||
|
||||
for (uint32_t c = 0; c < aOutputChannelCount; ++c) {
|
||||
aOutputChannels[c][s] = outputChannels[c];
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
} // namespace mozilla
|
||||
|
||||
|
|
|
@ -206,13 +206,13 @@ AudioSegment::Mix(AudioMixer& aMixer, uint32_t aOutputChannels,
|
|||
AudioDataValue* ptr =
|
||||
PointerForOffsetInChannel(buf.Elements(), outBufferLength,
|
||||
aOutputChannels, channel, offsetSamples);
|
||||
PodCopy(ptr, reinterpret_cast<const float*>(channelData[channel]),
|
||||
PodCopy(ptr, reinterpret_cast<const AudioDataValue*>(channelData[channel]),
|
||||
frames);
|
||||
}
|
||||
MOZ_ASSERT(channelData.Length() == aOutputChannels);
|
||||
} else if (channelData.Length() > aOutputChannels) {
|
||||
// Down mix.
|
||||
nsAutoTArray<float*, GUESS_AUDIO_CHANNELS> outChannelPtrs;
|
||||
nsAutoTArray<AudioDataValue*, GUESS_AUDIO_CHANNELS> outChannelPtrs;
|
||||
outChannelPtrs.SetLength(aOutputChannels);
|
||||
uint32_t offsetSamples = 0;
|
||||
for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
|
||||
|
@ -228,7 +228,7 @@ AudioSegment::Mix(AudioMixer& aMixer, uint32_t aOutputChannels,
|
|||
AudioDataValue* ptr =
|
||||
PointerForOffsetInChannel(buf.Elements(), outBufferLength,
|
||||
aOutputChannels, channel, offsetSamples);
|
||||
PodCopy(ptr, reinterpret_cast<const float*>(channelData[channel]),
|
||||
PodCopy(ptr, reinterpret_cast<const AudioDataValue*>(channelData[channel]),
|
||||
frames);
|
||||
}
|
||||
}
|
||||
|
|
Загрузка…
Ссылка в новой задаче