Drop WebRTC audio data if OS has skipped frames.

content/renderer/media/webrtc_audio_renderer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include <utility>
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
@@ skipping 169 matching lines @@
     const url::Origin& security_origin)
     : state_(UNINITIALIZED),
       source_render_frame_id_(source_render_frame_id),
       session_id_(session_id),
       signaling_thread_(signaling_thread),
       media_stream_(media_stream),
       source_(NULL),
       play_ref_count_(0),
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
-      fifo_delay_milliseconds_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                    media::CHANNEL_LAYOUT_STEREO,
                    0,
                    16,
                    0),
       output_device_id_(device_id),
       security_origin_(security_origin),
       render_callback_count_(0) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_frame_id=%d, session_id=%d, effects=%i",
@@ skipping 226 matching lines @@
   base::AutoLock auto_lock(lock_);
   if (!source_)
     return 0;
 
   DVLOG(2) << "WebRtcAudioRenderer::Render()";
   DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
 
   DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX));
   audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
 
+  // If there are skipped frames, pull and throw away the same amount.
+  if (frames_skipped > 0) {
+    const uint32_t frames_per_10ms =
+        static_cast<uint32_t>(sink_params_.sample_rate() / 100);
+    if (!audio_fifo_ && frames_skipped != frames_per_10ms) {
+      audio_fifo_.reset(new media::AudioPullFifo(
+          sink_params_.channels(), frames_per_10ms,
+          base::Bind(&WebRtcAudioRenderer::SourceCallback,
+                     base::Unretained(this))));
+    }
+
+    scoped_ptr<media::AudioBus> drop_bus =
+        media::AudioBus::Create(audio_bus->channels(), frames_skipped);
+    if (audio_fifo_)
+      audio_fifo_->Consume(drop_bus.get(), drop_bus->frames());
+    else
+      SourceCallback(0, drop_bus.get());
+  }
+
+  // Pull the data we shall deliver.

[tommi (sloooow) - chröme, 2016/01/15 18:50:03]

nit: s/shall/will

[Henrik Grunell, 2016/01/18 13:00:54]

Done.

   if (audio_fifo_)
     audio_fifo_->Consume(audio_bus, audio_bus->frames());
   else
     SourceCallback(0, audio_bus);
 
   return (state_ == PLAYING) ? audio_bus->frames() : 0;
 }
 
 void WebRtcAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
   LOG(ERROR) << "OnRenderError()";
 }
 
 // Called by AudioPullFifo when more data is necessary.
 void WebRtcAudioRenderer::SourceCallback(
     int fifo_frame_delay, media::AudioBus* audio_bus) {
   DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
   base::TimeTicks start_time = base::TimeTicks::Now();
   DVLOG(2) << "WebRtcAudioRenderer::SourceCallback("
            << fifo_frame_delay << ", "
            << audio_bus->frames() << ")";
 
   int output_delay_milliseconds = audio_delay_milliseconds_;
-  output_delay_milliseconds += fifo_delay_milliseconds_;
+  output_delay_milliseconds += fifo_frame_delay *
+                               base::Time::kMillisecondsPerSecond /
+                               sink_params_.sample_rate();
   DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds;
 
   // We need to keep render data for the |source_| regardless of |state_|,
   // otherwise the data will be buffered up inside |source_|.
   source_->RenderData(audio_bus, sink_params_.sample_rate(),
                       output_delay_milliseconds,
                       &current_time_);
 
   // Avoid filling up the audio bus if we are not playing; instead
   // return here and ensure that the returned value in Render() is 0.
@@ skipping 149 matching lines @@
   const int frames_per_buffer = GetOptimalBufferSize(
       sample_rate, sink_->GetOutputParameters().frames_per_buffer());
 
   new_sink_params.Reset(
       new_sink_params.format(), new_sink_params.channel_layout(),
       sample_rate, 16, frames_per_buffer);
 
   // Create a FIFO if re-buffering is required to match the source input with
   // the sink request. The source acts as provider here and the sink as
   // consumer.
-  int new_fifo_delay_milliseconds = 0;
   scoped_ptr<media::AudioPullFifo> new_audio_fifo;
-  if (source_params.frames_per_buffer() !=
-      new_sink_params.frames_per_buffer()) {
+  DCHECK_EQ(source_params.channels(),
+            media::ChannelLayoutToChannelCount(media::CHANNEL_LAYOUT_STEREO));

[tommi (sloooow) - chröme, 2016/01/15 18:50:03]

new assumption or did the previous code assume this?

[Henrik Grunell, 2016/01/18 13:00:54]

Actually, the assumption is that source and sink operates on the same. But the
source params is always initialized with the sink params' channel layout. In
addition, the sink layout is never changed. So this assert is unnecessary.
Though the code in this function is a bit hard to read. Removed.

+  const bool different_source_sink_frames =
+      source_params.frames_per_buffer() != new_sink_params.frames_per_buffer();
+  if (different_source_sink_frames) {
     DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
              << " to " << new_sink_params.frames_per_buffer();
-    new_audio_fifo.reset(new media::AudioPullFifo(
-        source_params.channels(), source_params.frames_per_buffer(),
-        base::Bind(&WebRtcAudioRenderer::SourceCallback,
-                   base::Unretained(this))));
-
-    if (new_sink_params.frames_per_buffer() >
-        source_params.frames_per_buffer()) {
-      int frame_duration_milliseconds =
-          base::Time::kMillisecondsPerSecond /
-          static_cast<double>(source_params.sample_rate());
-      new_fifo_delay_milliseconds = (new_sink_params.frames_per_buffer() -
-                                     source_params.frames_per_buffer()) *
-                                    frame_duration_milliseconds;
-    }
   }
 
   {
     base::AutoLock lock(lock_);
+    const bool source_frames_changed =
+        audio_fifo_->SizeInFrames() != source_params.frames_per_buffer();
+    if ((!audio_fifo_ && different_source_sink_frames) ||
+        (audio_fifo_ && source_frames_changed)) {
+      audio_fifo_.reset(new media::AudioPullFifo(
+          source_params.channels(), source_params.frames_per_buffer(),
+          base::Bind(&WebRtcAudioRenderer::SourceCallback,
+                     base::Unretained(this))));
+    }
     sink_params_ = new_sink_params;
-    fifo_delay_milliseconds_ = new_fifo_delay_milliseconds;
-    if (new_audio_fifo.get())
-      audio_fifo_ = std::move(new_audio_fifo);
   }
 
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content