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

Issue 857093002: Set audio format before adding pepper sink to audio track. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Fix tests. Created 5 years, 11 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_track.h" 5 #include "content/renderer/media/webrtc_local_audio_track.h"
6 6
7 #include <limits> 7 #include <limits>
8 8
9 #include "content/public/renderer/media_stream_audio_sink.h" 9 #include "content/public/renderer/media_stream_audio_sink.h"
10 #include "content/renderer/media/media_stream_audio_level_calculator.h" 10 #include "content/renderer/media/media_stream_audio_level_calculator.h"
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33 DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()"; 33 DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
34 } 34 }
35 35
36 WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { 36 WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
37 DCHECK(main_render_thread_checker_.CalledOnValidThread()); 37 DCHECK(main_render_thread_checker_.CalledOnValidThread());
38 DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()"; 38 DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
39 // Users might not call Stop() on the track. 39 // Users might not call Stop() on the track.
40 Stop(); 40 Stop();
41 } 41 }
42 42
43 const media::AudioParameters& WebRtcLocalAudioTrack::GetOutputFormat() const {
44 if (webaudio_source_.get()) {
45 static const media::AudioParameters empty_params;
dmichael (off chromium) 2015/01/20 17:07:03 ditto... it's almost always better to just return
DaleCurtis 2015/01/20 18:46:56 +1, just return media::AudioParameters().
Anand Mistry (off Chromium) 2015/01/27 02:12:10 Done. I just figured returning const reference was
dmichael (off chromium) 2015/01/27 18:24:10 Copying a small-ish object that doesn't do any hea
46 return empty_params;
47 } else {
48 return capturer_->GetOutputFormat();
49 }
50 }
51
43 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus, 52 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
44 base::TimeTicks estimated_capture_time, 53 base::TimeTicks estimated_capture_time,
45 bool force_report_nonzero_energy) { 54 bool force_report_nonzero_energy) {
46 DCHECK(capture_thread_checker_.CalledOnValidThread()); 55 DCHECK(capture_thread_checker_.CalledOnValidThread());
47 DCHECK(!estimated_capture_time.is_null()); 56 DCHECK(!estimated_capture_time.is_null());
48 57
49 // Calculate the signal level regardless of whether the track is disabled or 58 // Calculate the signal level regardless of whether the track is disabled or
50 // enabled. If |force_report_nonzero_energy| is true, |audio_bus| contains 59 // enabled. If |force_report_nonzero_energy| is true, |audio_bus| contains
51 // post-processed data that may be all zeros even though the signal contained 60 // post-processed data that may be all zeros even though the signal contained
52 // energy before the processing. In this case, report nonzero energy even if 61 // energy before the processing. In this case, report nonzero energy even if
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221 (*it)->Reset(); 230 (*it)->Reset();
222 } 231 }
223 } 232 }
224 233
225 webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() { 234 webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
226 DCHECK(thread_checker_.CalledOnValidThread()); 235 DCHECK(thread_checker_.CalledOnValidThread());
227 return adapter_.get(); 236 return adapter_.get();
228 } 237 }
229 238
230 } // namespace content 239 } // namespace content
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