Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1232)

Side by Side Diff: webrtc/audio/audio_receive_stream.cc

Issue 1991233004: Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@audio-decoder-factory-injections-3
Patch Set: Parental Advisory: Explicit Content Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/audio/DEPS ('k') | webrtc/audio/audio_receive_stream_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 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
(...skipping 76 matching lines...) Expand 10 before | Expand all | Expand 10 after
87 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); 87 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
88 RTC_DCHECK_NE(config_.voe_channel_id, -1); 88 RTC_DCHECK_NE(config_.voe_channel_id, -1);
89 RTC_DCHECK(audio_state_.get()); 89 RTC_DCHECK(audio_state_.get());
90 RTC_DCHECK(congestion_controller); 90 RTC_DCHECK(congestion_controller);
91 RTC_DCHECK(rtp_header_parser_); 91 RTC_DCHECK(rtp_header_parser_);
92 92
93 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); 93 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
94 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); 94 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
95 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); 95 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
96 96
97 // TODO(ossu): This is where we'd like to set the decoder factory to
98 // use. However, since it needs to be included when constructing Channel, we
99 // cannot do that until we're able to move Channel ownership into the
100 // Audio{Send,Receive}Streams. The best we can do is check that we're not
101 // trying to use two different factories using the different interfaces.
102 RTC_CHECK(config.decoder_factory);
103 RTC_CHECK_EQ(config.decoder_factory,
104 channel_proxy_->GetAudioDecoderFactory());
105
97 channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport); 106 channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
98 107
99 for (const auto& extension : config.rtp.extensions) { 108 for (const auto& extension : config.rtp.extensions) {
100 if (extension.uri == RtpExtension::kAudioLevelUri) { 109 if (extension.uri == RtpExtension::kAudioLevelUri) {
101 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); 110 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
102 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( 111 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
103 kRtpExtensionAudioLevel, extension.id); 112 kRtpExtensionAudioLevel, extension.id);
104 RTC_DCHECK(registered); 113 RTC_DCHECK(registered);
105 } else if (extension.uri == RtpExtension::kAbsSendTimeUri) { 114 } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
106 channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id); 115 channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
(...skipping 136 matching lines...) Expand 10 before | Expand all | Expand 10 after
243 252
244 VoiceEngine* AudioReceiveStream::voice_engine() const { 253 VoiceEngine* AudioReceiveStream::voice_engine() const {
245 internal::AudioState* audio_state = 254 internal::AudioState* audio_state =
246 static_cast<internal::AudioState*>(audio_state_.get()); 255 static_cast<internal::AudioState*>(audio_state_.get());
247 VoiceEngine* voice_engine = audio_state->voice_engine(); 256 VoiceEngine* voice_engine = audio_state->voice_engine();
248 RTC_DCHECK(voice_engine); 257 RTC_DCHECK(voice_engine);
249 return voice_engine; 258 return voice_engine;
250 } 259 }
251 } // namespace internal 260 } // namespace internal
252 } // namespace webrtc 261 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/audio/DEPS ('k') | webrtc/audio/audio_receive_stream_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698