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Bug 901633 - Part 10 - Change the receiving side of the MediaPipeline so that it can detect and handle stereo. r=jesup

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--HG--
extra : rebase_source : ec1f808a2417524d484821656430c21268dd05a7
This commit is contained in:
Paul Adenot 2015-08-12 14:43:51 +02:00
parent a84ad7251d
commit db0807f658
1 changed files with 35 additions and 12 deletions
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47
media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
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@ -1330,18 +1330,18 @@ NotifyPull(MediaStreamGraph* graph, StreamTime desired_time) {
while (source_->TicksToTimeRoundDown(track_rate_, played_ticks_) < while (source_->TicksToTimeRoundDown(track_rate_, played_ticks_) <
desired_time) { desired_time) {
// TODO(ekr@rtfm.com): Is there a way to avoid mallocating here? Or reduce the size? // TODO(ekr@rtfm.com): Is there a way to avoid mallocating here? Or reduce the size?
// Max size given mono is 480*2*1 = 960 (48KHz) // Max size given stereo is 480*2*2 = 1920 (48KHz)
#define AUDIO_SAMPLE_BUFFER_MAX 1000 const size_t AUDIO_SAMPLE_BUFFER_MAX = 1920;
MOZ_ASSERT((track_rate_/100)*sizeof(uint16_t) <= AUDIO_SAMPLE_BUFFER_MAX); MOZ_ASSERT((track_rate_/100)*sizeof(uint16_t) * 2 <= AUDIO_SAMPLE_BUFFER_MAX);
int16_t scratch_buffer[AUDIO_SAMPLE_BUFFER_MAX];
nsRefPtr<SharedBuffer> samples = SharedBuffer::Create(AUDIO_SAMPLE_BUFFER_MAX);
int16_t *samples_data = static_cast<int16_t *>(samples->Data());
int samples_length; int samples_length;
// This fetches 10ms of data // This fetches 10ms of data, either mono or stereo
MediaConduitErrorCode err = MediaConduitErrorCode err =
static_cast<AudioSessionConduit*>(conduit_.get())->GetAudioFrame( static_cast<AudioSessionConduit*>(conduit_.get())->GetAudioFrame(
samples_data, scratch_buffer,
track_rate_, track_rate_,
0, // TODO(ekr@rtfm.com): better estimate of "capture" (really playout) delay 0, // TODO(ekr@rtfm.com): better estimate of "capture" (really playout) delay
samples_length); samples_length);
@ -1353,7 +1353,7 @@ NotifyPull(MediaStreamGraph* graph, StreamTime desired_time) {
<< " (desired " << desired_time << " -> " << " (desired " << desired_time << " -> "
<< source_->StreamTimeToSeconds(desired_time) << ")"); << source_->StreamTimeToSeconds(desired_time) << ")");
samples_length = track_rate_/100; // if this is not enough we'll loop and provide more samples_length = track_rate_/100; // if this is not enough we'll loop and provide more
memset(samples_data, '\0', samples_length * sizeof(uint16_t)); PodArrayZero(scratch_buffer);
} }
MOZ_ASSERT(samples_length * sizeof(uint16_t) < AUDIO_SAMPLE_BUFFER_MAX); MOZ_ASSERT(samples_length * sizeof(uint16_t) < AUDIO_SAMPLE_BUFFER_MAX);
@ -1361,14 +1361,37 @@ NotifyPull(MediaStreamGraph* graph, StreamTime desired_time) {
MOZ_MTLOG(ML_DEBUG, "Audio conduit returned buffer of length " MOZ_MTLOG(ML_DEBUG, "Audio conduit returned buffer of length "
<< samples_length); << samples_length);
nsRefPtr<SharedBuffer> samples = SharedBuffer::Create(samples_length * sizeof(uint16_t));
int16_t *samples_data = static_cast<int16_t *>(samples->Data());
AudioSegment segment; AudioSegment segment;
nsAutoTArray<const int16_t*,1> channels; // We derive the number of channels of the stream from the number of samples
channels.AppendElement(samples_data); // the AudioConduit gives us, considering it gives us packets of 10ms and we
segment.AppendFrames(samples.forget(), channels, samples_length); // know the rate.
uint32_t channelCount = samples_length / (track_rate_ / 100);
nsAutoTArray<int16_t*,2> channels;
nsAutoTArray<const int16_t*,2> outputChannels;
size_t frames = samples_length / channelCount;
channels.SetLength(channelCount);
size_t offset = 0;
for (size_t i = 0; i < channelCount; i++) {
channels[i] = samples_data + offset;
offset += frames;
}
DeinterleaveAndConvertBuffer(scratch_buffer,
frames,
channelCount,
channels.Elements());
outputChannels.AppendElements(channels);
segment.AppendFrames(samples.forget(), outputChannels, frames);
// Handle track not actually added yet or removed/finished // Handle track not actually added yet or removed/finished
if (source_->AppendToTrack(track_id_, &segment)) { if (source_->AppendToTrack(track_id_, &segment)) {
played_ticks_ += track_rate_/100; // 10ms in TrackTicks played_ticks_ += frames;
} else { } else {
MOZ_MTLOG(ML_ERROR, "AppendToTrack failed"); MOZ_MTLOG(ML_ERROR, "AppendToTrack failed");
// we can't un-read the data, but that's ok since we don't want to // we can't un-read the data, but that's ok since we don't want to