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Issue 2620303003: Replace ASSERT by RTC_DCHECK in all non-test code. (Closed)
Patch Set: Address final nits. Created 3 years, 11 months ago
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1 /* 1 /*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "webrtc/api/rtpsender.h" 11 #include "webrtc/api/rtpsender.h"
12 12
13 #include "webrtc/api/localaudiosource.h" 13 #include "webrtc/api/localaudiosource.h"
14 #include "webrtc/api/mediastreaminterface.h" 14 #include "webrtc/api/mediastreaminterface.h"
15 #include "webrtc/base/checks.h"
15 #include "webrtc/base/helpers.h" 16 #include "webrtc/base/helpers.h"
16 #include "webrtc/base/trace_event.h" 17 #include "webrtc/base/trace_event.h"
17 18
18 namespace webrtc { 19 namespace webrtc {
19 20
20 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {} 21 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
21 22
22 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() { 23 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
23 rtc::CritScope lock(&lock_); 24 rtc::CritScope lock(&lock_);
24 if (sink_) 25 if (sink_)
25 sink_->OnClose(); 26 sink_->OnClose();
26 } 27 }
27 28
28 void LocalAudioSinkAdapter::OnData(const void* audio_data, 29 void LocalAudioSinkAdapter::OnData(const void* audio_data,
29 int bits_per_sample, 30 int bits_per_sample,
30 int sample_rate, 31 int sample_rate,
31 size_t number_of_channels, 32 size_t number_of_channels,
32 size_t number_of_frames) { 33 size_t number_of_frames) {
33 rtc::CritScope lock(&lock_); 34 rtc::CritScope lock(&lock_);
34 if (sink_) { 35 if (sink_) {
35 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, 36 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
36 number_of_frames); 37 number_of_frames);
37 } 38 }
38 } 39 }
39 40
40 void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) { 41 void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
41 rtc::CritScope lock(&lock_); 42 rtc::CritScope lock(&lock_);
42 ASSERT(!sink || !sink_); 43 RTC_DCHECK(!sink || !sink_);
43 sink_ = sink; 44 sink_ = sink;
44 } 45 }
45 46
46 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track, 47 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
47 const std::string& stream_id, 48 const std::string& stream_id,
48 cricket::VoiceChannel* channel, 49 cricket::VoiceChannel* channel,
49 StatsCollector* stats) 50 StatsCollector* stats)
50 : id_(track->id()), 51 : id_(track->id()),
51 stream_id_(stream_id), 52 stream_id_(stream_id),
52 channel_(channel), 53 channel_(channel),
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399 LOG(LS_WARNING) << "SetVideoSend: No video channel exists."; 400 LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
400 return; 401 return;
401 } 402 }
402 // Allow SetVideoSend to fail since |enable| is false and |source| is null. 403 // Allow SetVideoSend to fail since |enable| is false and |source| is null.
403 // This the normal case when the underlying media channel has already been 404 // This the normal case when the underlying media channel has already been
404 // deleted. 405 // deleted.
405 channel_->SetVideoSend(ssrc_, false, nullptr, nullptr); 406 channel_->SetVideoSend(ssrc_, false, nullptr, nullptr);
406 } 407 }
407 408
408 } // namespace webrtc 409 } // namespace webrtc
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