Revert of Revert of Remove the last piece of deprecated synchronous IO code.

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 "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 200 matching lines @@
       source_render_view_id_(source_render_view_id),
       source_render_frame_id_(source_render_frame_id),
       session_id_(session_id),
       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, sample_rate, 16,
+                   media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
                    frames_per_buffer,
                    GetCurrentDuckingFlag(source_render_frame_id)) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_view_id=%d"
       ", session_id=%d, sample_rate=%d, frames_per_buffer=%d, effects=%i",
       source_render_view_id,
       session_id,
       sample_rate,
       frames_per_buffer,
       sink_params_.effects()));
@@ skipping 46 matching lines @@
 
   // Set up audio parameters for the source, i.e., the WebRTC client.
 
   // The WebRTC client only supports multiples of 10ms as buffer size where
   // 10ms is preferred for lowest possible delay.
   media::AudioParameters source_params;
   const int frames_per_10ms = (sample_rate / 100);
   DVLOG(1) << "Using WebRTC output buffer size: " << frames_per_10ms;
 
   source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                      sink_params_.channel_layout(), sink_params_.channels(), 0,
+                      sink_params_.channel_layout(), sink_params_.channels(),
                       sample_rate, 16, frames_per_10ms);
 
   // Update audio parameters for the sink, i.e., the native audio output stream.
   // We strive to open up using native parameters to achieve best possible
   // performance and to ensure that no FIFO is needed on the browser side to
   // match the client request. Any mismatch between the source and the sink is
   // taken care of in this class instead using a pull FIFO.
 
   // Use native output size as default.
   int frames_per_buffer = sink_params_.frames_per_buffer();
 #if defined(OS_ANDROID)
   // TODO(henrika): Keep tuning this scheme and espcicially for low-latency
   // cases. Might not be possible to come up with the perfect solution using
   // the render side only.
   if (frames_per_buffer < 2 * frames_per_10ms) {
     // Examples of low-latency frame sizes and the resulting |buffer_size|:
     //  Nexus 7     : 240 audio frames => 2*480 = 960
     //  Nexus 10    : 256              => 2*441 = 882
     //  Galaxy Nexus: 144              => 2*441 = 882
     frames_per_buffer = 2 * frames_per_10ms;
     DVLOG(1) << "Low-latency output detected on Android";
   }
 #endif
   DVLOG(1) << "Using sink output buffer size: " << frames_per_buffer;
 
   sink_params_.Reset(sink_params_.format(), sink_params_.channel_layout(),
-                     sink_params_.channels(), 0, sample_rate, 16,
+                     sink_params_.channels(), 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.
   fifo_delay_milliseconds_ = 0;
   if (source_params.frames_per_buffer() != sink_params_.frames_per_buffer()) {
     DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
              << " to " << sink_params_.frames_per_buffer();
     audio_fifo_.reset(new media::AudioPullFifo(
@@ skipping 277 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content