Refactor AudioParameters member setting.

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 215 matching lines @@
         "WebRTC.AudioOutputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected",
                          sample_rate);
   }
 
   // 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(),
-                      sample_rate, 16, frames_per_10ms);
+  media::AudioParameters source_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      sink_params_.channel_layout(), sample_rate, 16, frames_per_10ms);
+  source_params.set_channels_for_discrete(sink_params_.channels());
 
   const int frames_per_buffer =
       GetOptimalBufferSize(sample_rate, sink_params_.frames_per_buffer());
 
   sink_params_.Reset(sink_params_.format(), sink_params_.channel_layout(),
-                     sink_params_.channels(), sample_rate, 16,
-                     frames_per_buffer);
+                     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(
         source_params.channels(),
@@ skipping 173 matching lines @@
 }
 
 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) {
-  base::TimeTicks start_time = base::TimeTicks::Now() ;
+  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_;
   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_|.
@@ skipping 99 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content