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

Issue 503683003: Remove implicit conversions from scoped_refptr to T* in content/renderer/media/webrtc* (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 3 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_local_audio_renderer.h" 5 #include "content/renderer/media/webrtc_local_audio_renderer.h"
6 6
7 #include "base/debug/trace_event.h" 7 #include "base/debug/trace_event.h"
8 #include "base/logging.h" 8 #include "base/logging.h"
9 #include "base/message_loop/message_loop_proxy.h" 9 #include "base/message_loop/message_loop_proxy.h"
10 #include "base/metrics/histogram.h" 10 #include "base/metrics/histogram.h"
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146 146
147 { 147 {
148 base::AutoLock auto_lock(thread_lock_); 148 base::AutoLock auto_lock(thread_lock_);
149 playing_ = false; 149 playing_ = false;
150 loopback_fifo_.reset(); 150 loopback_fifo_.reset();
151 } 151 }
152 152
153 // Stop the output audio stream, i.e, stop asking for data to render. 153 // Stop the output audio stream, i.e, stop asking for data to render.
154 // It is safer to call Stop() on the |sink_| to clean up the resources even 154 // It is safer to call Stop() on the |sink_| to clean up the resources even
155 // when the |sink_| is never started. 155 // when the |sink_| is never started.
156 if (sink_) { 156 if (sink_.get()) {
157 sink_->Stop(); 157 sink_->Stop();
158 sink_ = NULL; 158 sink_ = NULL;
159 } 159 }
160 160
161 if (!sink_started_) { 161 if (!sink_started_) {
162 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", 162 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
163 kSinkNeverStarted, kSinkStatesMax); 163 kSinkNeverStarted, kSinkStatesMax);
164 } 164 }
165 sink_started_ = false; 165 sink_started_ = false;
166 166
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309 // the FIFO overflows. 309 // the FIFO overflows.
310 const int blocks_of_buffers = 310 const int blocks_of_buffers =
311 10 * params.frames_per_buffer() / sink_params_.frames_per_buffer() + 1; 311 10 * params.frames_per_buffer() / sink_params_.frames_per_buffer() + 1;
312 media::AudioBlockFifo* new_fifo = new media::AudioBlockFifo( 312 media::AudioBlockFifo* new_fifo = new media::AudioBlockFifo(
313 params.channels(), sink_params_.frames_per_buffer(), blocks_of_buffers); 313 params.channels(), sink_params_.frames_per_buffer(), blocks_of_buffers);
314 314
315 base::AutoLock auto_lock(thread_lock_); 315 base::AutoLock auto_lock(thread_lock_);
316 loopback_fifo_.reset(new_fifo); 316 loopback_fifo_.reset(new_fifo);
317 } 317 }
318 318
319 if (!sink_) 319 if (!sink_.get())
320 return; // WebRtcLocalAudioRenderer has not yet been started. 320 return; // WebRtcLocalAudioRenderer has not yet been started.
321 321
322 // Stop |sink_| and re-create a new one to be initialized with different audio 322 // Stop |sink_| and re-create a new one to be initialized with different audio
323 // parameters. Then, invoke MaybeStartSink() to restart everything again. 323 // parameters. Then, invoke MaybeStartSink() to restart everything again.
324 if (sink_started_) { 324 if (sink_started_) {
325 sink_->Stop(); 325 sink_->Stop();
326 sink_started_ = false; 326 sink_started_ = false;
327 } 327 }
328 328
329 sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_, 329 sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_,
330 source_render_frame_id_); 330 source_render_frame_id_);
331 MaybeStartSink(); 331 MaybeStartSink();
332 } 332 }
333 333
334 } // namespace content 334 } // namespace content
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