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Side by Side Diff: content/renderer/media/webrtc_audio_renderer.cc

Issue 293673004: Remove unused RenderIO() interface. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: Created 6 years, 7 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "content/renderer/media/webrtc_audio_renderer.h" 5 #include "content/renderer/media/webrtc_audio_renderer.h"
6 6
7 #include "base/logging.h" 7 #include "base/logging.h"
8 #include "base/metrics/histogram.h" 8 #include "base/metrics/histogram.h"
9 #include "base/strings/string_util.h" 9 #include "base/strings/string_util.h"
10 #include "base/strings/stringprintf.h" 10 #include "base/strings/stringprintf.h"
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317 source_params.frames_per_buffer()) * frame_duration_milliseconds; 317 source_params.frames_per_buffer()) * frame_duration_milliseconds;
318 } 318 }
319 } 319 }
320 320
321 source_ = source; 321 source_ = source;
322 322
323 // Configure the audio rendering client and start rendering. 323 // Configure the audio rendering client and start rendering.
324 sink_ = AudioDeviceFactory::NewOutputDevice( 324 sink_ = AudioDeviceFactory::NewOutputDevice(
325 source_render_view_id_, source_render_frame_id_); 325 source_render_view_id_, source_render_frame_id_);
326 326
327 // TODO(tommi): Rename InitializeUnifiedStream to rather reflect association
328 // with a session.
329 DCHECK_GE(session_id_, 0); 327 DCHECK_GE(session_id_, 0);
330 sink_->InitializeUnifiedStream(sink_params_, this, session_id_); 328 sink_->InitializeWithSessionId(sink_params_, this, session_id_);
331 329
332 sink_->Start(); 330 sink_->Start();
333 331
334 // User must call Play() before any audio can be heard. 332 // User must call Play() before any audio can be heard.
335 state_ = PAUSED; 333 state_ = PAUSED;
336 334
337 UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer", 335 UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
338 source_params.frames_per_buffer(), 336 source_params.frames_per_buffer(),
339 kUnexpectedAudioBufferSize); 337 kUnexpectedAudioBufferSize);
340 AddHistogramFramesPerBuffer(source_params.frames_per_buffer()); 338 AddHistogramFramesPerBuffer(source_params.frames_per_buffer());
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583 if (RemovePlayingState(source, state)) 581 if (RemovePlayingState(source, state))
584 EnterPauseState(); 582 EnterPauseState();
585 } else if (AddPlayingState(source, state)) { 583 } else if (AddPlayingState(source, state)) {
586 EnterPlayState(); 584 EnterPlayState();
587 } 585 }
588 UpdateSourceVolume(source); 586 UpdateSourceVolume(source);
589 } 587 }
590 } 588 }
591 589
592 } // namespace content 590 } // namespace content
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