Index: content/renderer/media/webaudio_capturer_source.cc |
diff --git a/content/renderer/media/webaudio_capturer_source.cc b/content/renderer/media/webaudio_capturer_source.cc |
index bcebcbb3e24e523288827db01d6b740a019a29fd..6a4966070870b38b1e0627cb6aeca2e93da426c2 100644 |
--- a/content/renderer/media/webaudio_capturer_source.cc |
+++ b/content/renderer/media/webaudio_capturer_source.cc |
@@ -4,27 +4,28 @@ |
#include "content/renderer/media/webaudio_capturer_source.h" |
+#include "base/bind.h" |
+#include "base/bind_helpers.h" |
#include "base/logging.h" |
#include "base/time/time.h" |
#include "content/renderer/media/webrtc_local_audio_track.h" |
using media::AudioBus; |
-using media::AudioFifo; |
using media::AudioParameters; |
using media::ChannelLayout; |
using media::CHANNEL_LAYOUT_MONO; |
using media::CHANNEL_LAYOUT_STEREO; |
-static const int kMaxNumberOfBuffersInFifo = 5; |
- |
namespace content { |
WebAudioCapturerSource::WebAudioCapturerSource( |
const blink::WebMediaStreamSource& blink_source) |
: track_(NULL), |
audio_format_changed_(false), |
- blink_source_(blink_source) { |
-} |
+ rechunker_(base::TimeDelta::FromMilliseconds(10), |
+ base::Bind(&WebAudioCapturerSource::DeliverRechunkedAudio, |
+ base::Unretained(this))), |
+ blink_source_(blink_source) {} |
WebAudioCapturerSource::~WebAudioCapturerSource() { |
DCHECK(thread_checker_.CalledOnValidThread()); |
@@ -48,20 +49,19 @@ void WebAudioCapturerSource::setFormat( |
// Set the format used by this WebAudioCapturerSource. We are using 10ms data |
// as buffer size since that is the native buffer size of WebRtc packet |
// running on. |
+ rechunker_.SetSampleRate(sample_rate); |
params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, |
- sample_rate, 16, sample_rate / 100); |
+ sample_rate, 16, rechunker_.output_frames()); |
// Take care of the discrete channel layout case. |
params_.set_channels_for_discrete(number_of_channels); |
audio_format_changed_ = true; |
- wrapper_bus_ = AudioBus::CreateWrapper(params_.channels()); |
- capture_bus_ = AudioBus::Create(params_); |
- |
- fifo_.reset(new AudioFifo( |
- params_.channels(), |
- kMaxNumberOfBuffersInFifo * params_.frames_per_buffer())); |
+ if (!wrapper_bus_ || |
+ wrapper_bus_->channels() != static_cast<int>(number_of_channels)) { |
+ wrapper_bus_ = AudioBus::CreateWrapper(params_.channels()); |
+ } |
} |
void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) { |
@@ -85,6 +85,11 @@ void WebAudioCapturerSource::Stop() { |
void WebAudioCapturerSource::consumeAudio( |
const blink::WebVector<const float*>& audio_data, |
size_t number_of_frames) { |
+ // TODO(miu): Plumbing is needed to determine the actual capture timestamp |
+ // of the audio, instead of just snapshotting TimeTicks::Now(), for proper |
+ // audio/video sync. http://crbug.com/335335 |
+ base::TimeTicks reference_time = base::TimeTicks::Now(); |
+ |
base::AutoLock auto_lock(lock_); |
if (!track_) |
return; |
@@ -96,36 +101,20 @@ void WebAudioCapturerSource::consumeAudio( |
} |
wrapper_bus_->set_frames(number_of_frames); |
- |
- // Make sure WebKit is honoring what it told us up front |
- // about the channels. |
DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size())); |
- |
for (size_t i = 0; i < audio_data.size(); ++i) |
wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i])); |
- // Handle mismatch between WebAudio buffer-size and WebRTC. |
- int available = fifo_->max_frames() - fifo_->frames(); |
- if (available < static_cast<int>(number_of_frames)) { |
- NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun."; |
- return; |
- } |
- |
- // Compute the estimated capture time of the first sample frame of audio that |
- // will be consumed from the FIFO in the loop below. |
- base::TimeTicks estimated_capture_time = base::TimeTicks::Now() - |
- fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate(); |
- |
- fifo_->Push(wrapper_bus_.get()); |
- while (fifo_->frames() >= capture_bus_->frames()) { |
- fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames()); |
- track_->Capture(*capture_bus_, estimated_capture_time, false); |
+ // The following will result in zero, one, or multiple synchronous calls to |
+ // DeliverRechunkedAudio(). |
+ rechunker_.Push(*wrapper_bus_, reference_time - base::TimeTicks()); |
o1ka
2016/02/22 13:04:47
I like how neat it becomes!
miu
2016/02/23 04:27:41
Acknowledged.
|
+} |
- // Advance the estimated capture time for the next FIFO consume operation. |
- estimated_capture_time += |
- capture_bus_->frames() * base::TimeDelta::FromSeconds(1) / |
- params_.sample_rate(); |
- } |
+void WebAudioCapturerSource::DeliverRechunkedAudio( |
+ const media::AudioBus& audio_bus, |
+ base::TimeDelta reference_timestamp) { |
+ lock_.AssertAcquired(); |
+ track_->Capture(audio_bus, base::TimeTicks() + reference_timestamp, false); |
} |
// If registered as audio consumer in |blink_source_|, deregister from |