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Bug 1543622 - Make number of channels out param of GetAudioFrame; r=pehrsons

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The number of channels is available in mAudioFrame in GetAudioFrame so
there is no reason to calculate it after the fact in MediaPipeline.

Differential Revision: https://phabricator.services.mozilla.com/D50934

--HG--
extra : moz-landing-system : lando
This commit is contained in:
Dan Minor 2019-10-29 17:30:03 +00:00
parent 5d8c10a424
commit 1f3dd67cb9
5 changed files with 28 additions and 15 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
media/webrtc/signaling/gtest/mediaconduit_unittests.cpp
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@ -197,7 +197,8 @@ void AudioSendAndReceive::GenerateAndReadSamples() {
auto audioInput = mozilla::MakeUnique<int16_t[]>(PLAYOUT_SAMPLE_LENGTH);
auto audioOutput = mozilla::MakeUnique<int16_t[]>(PLAYOUT_SAMPLE_LENGTH);
short* inbuf;
int sampleLengthDecoded = 0;
size_t numChannels = 0;
size_t sampleLengthDecoded = 0;
unsigned int SAMPLES = (PLAYOUT_SAMPLE_FREQUENCY / 100); // 10 milliseconds
int CHANNELS = 1; // mono audio
int sampleLengthInBytes = sizeof(int16_t) * PLAYOUT_SAMPLE_LENGTH;
@ -246,7 +247,7 @@ void AudioSendAndReceive::GenerateAndReadSamples() {
PR_Sleep(PR_MillisecondsToInterval(10));
mOtherSession->GetAudioFrame(audioOutput.get(), PLAYOUT_SAMPLE_FREQUENCY,
10, sampleLengthDecoded);
10, numChannels, sampleLengthDecoded);
if (sampleLengthDecoded == 0) {
cerr << " Zero length Sample " << endl;
}

14
media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
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@ -583,7 +583,8 @@ MediaConduitErrorCode WebrtcAudioConduit::SendAudioFrame(
MediaConduitErrorCode WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
int32_t samplingFreqHz,
int32_t capture_delay,
int& lengthSamples) {
size_t& numChannels,
size_t& lengthSamples) {
CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
// validate params
@ -614,16 +615,23 @@ MediaConduitErrorCode WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
return kMediaConduitSessionNotInited;
}
int lengthSamplesAllowed = lengthSamples;
size_t lengthSamplesAllowed = lengthSamples;
lengthSamples = 0; // output paramter
mRecvChannelProxy->GetAudioFrameWithInfo(samplingFreqHz, &mAudioFrame);
numChannels = mAudioFrame.num_channels_;
if (numChannels == 0) {
CSFLogError(LOGTAG, "%s Audio frame has zero channels", __FUNCTION__);
return kMediaConduitPlayoutError;
}
// XXX Annoying, have to copy to our buffers -- refactor?
lengthSamples = mAudioFrame.samples_per_channel_ * mAudioFrame.num_channels_;
MOZ_RELEASE_ASSERT(lengthSamples <= lengthSamplesAllowed);
PodCopy(speechData, mAudioFrame.data(), lengthSamples);
CSFLogDebug(LOGTAG, "%s GetAudioFrame:Got samples: length %d ", __FUNCTION__,
CSFLogDebug(LOGTAG, "%s GetAudioFrame:Got samples: length %zu ", __FUNCTION__,
lengthSamples);
return kMediaConduitNoError;
}

5
media/webrtc/signaling/src/media-conduit/AudioConduit.h
View File

@ -137,6 +137,8 @@ class WebrtcAudioConduit : public AudioSessionConduit,
* @param capture_delay [in]: Estimated Time between reading of the samples
* to rendering/playback
* @param lengthSamples [in]: Contain maximum length of speechData array.
* @param numChannels [out]: Number of channels in the audio frame,
* guaranteed to be non-zero.
* @param lengthSamples [out]: Will contain length of the audio frame in
* samples at return.
* Ex: A value of 160 implies 160 samples each of
@ -151,7 +153,8 @@ class WebrtcAudioConduit : public AudioSessionConduit,
MediaConduitErrorCode GetAudioFrame(int16_t speechData[],
int32_t samplingFreqHz,
int32_t capture_delay,
int& lengthSamples) override;
size_t& numChannels,
size_t& lengthSamples) override;
/**
* Webrtc transport implementation to send and receive RTP packet.

5
media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
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@ -536,6 +536,8 @@ class AudioSessionConduit : public MediaSessionConduit {
* Hertz (16000, 32000,..)
* @param capture_delay [in]: Estimated Time between reading of the samples
* to rendering/playback
* @param numChannels [out]: Number of channels in the audio frame,
* guaranteed to be non-zero.
* @param lengthSamples [out]: Will contain length of the audio frame in
* samples at return.
* Ex: A value of 160 implies 160 samples each of
@ -549,7 +551,8 @@ class AudioSessionConduit : public MediaSessionConduit {
virtual MediaConduitErrorCode GetAudioFrame(int16_t speechData[],
int32_t samplingFreqHz,
int32_t capture_delay,
int& lengthSamples) = 0;
size_t& numChannels,
size_t& lengthSamples) = 0;
/**
* Checks if given sampling frequency is supported

14
media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
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@ -1320,11 +1320,12 @@ class MediaPipelineReceiveAudio::PipelineListener
// in the graph rate.
while (mPlayedTicks < aDesiredTime) {
const int scratchBufferLength =
constexpr size_t scratchBufferLength =
AUDIO_SAMPLE_BUFFER_MAX_BYTES / sizeof(int16_t);
int16_t scratchBuffer[scratchBufferLength];
int samplesLength = scratchBufferLength;
size_t channelCount = 0;
size_t samplesLength = scratchBufferLength;
// This fetches 10ms of data, either mono or stereo
MediaConduitErrorCode err =
@ -1332,7 +1333,7 @@ class MediaPipelineReceiveAudio::PipelineListener
->GetAudioFrame(scratchBuffer, mRate,
0, // TODO(ekr@rtfm.com): better estimate of
// "capture" (really playout) delay
samplesLength);
channelCount, samplesLength);
if (err != kMediaConduitNoError) {
// Insert silence on conduit/GIPS failure (extremely unlikely)
@ -1341,6 +1342,7 @@ class MediaPipelineReceiveAudio::PipelineListener
" (desired %" PRId64 " -> %f)",
err, mPlayedTicks, aDesiredTime,
mSource->TrackTimeToSeconds(aDesiredTime)));
channelCount = 1;
// if this is not enough we'll loop and provide more
samplesLength = samplesPer10ms;
PodArrayZero(scratchBuffer);
@ -1349,16 +1351,12 @@ class MediaPipelineReceiveAudio::PipelineListener
MOZ_RELEASE_ASSERT(samplesLength <= scratchBufferLength);
MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
("Audio conduit returned buffer of length %u", samplesLength));
("Audio conduit returned buffer of length %zu", samplesLength));
RefPtr<SharedBuffer> samples =
SharedBuffer::Create(samplesLength * sizeof(uint16_t));
int16_t* samplesData = static_cast<int16_t*>(samples->Data());
AudioSegment segment;
// We derive the number of channels of the stream from the number of
// samples the AudioConduit gives us, considering it gives us packets of
// 10ms and we know the rate.
uint32_t channelCount = samplesLength / samplesPer10ms;
AutoTArray<int16_t*, 2> channels;
AutoTArray<const int16_t*, 2> outputChannels;
size_t frames = samplesLength / channelCount;