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

Issue 1454073002: Move some receive stream configuration into webrtc::AudioReceiveStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 1 month ago
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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
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44 ss << "{rtp: " << rtp.ToString(); 44 ss << "{rtp: " << rtp.ToString();
45 ss << ", voe_channel_id: " << voe_channel_id; 45 ss << ", voe_channel_id: " << voe_channel_id;
46 // TODO(solenberg): Encoder config. 46 // TODO(solenberg): Encoder config.
47 ss << ", cng_payload_type: " << cng_payload_type; 47 ss << ", cng_payload_type: " << cng_payload_type;
48 ss << ", red_payload_type: " << red_payload_type; 48 ss << ", red_payload_type: " << red_payload_type;
49 ss << '}'; 49 ss << '}';
50 return ss.str(); 50 return ss.str();
51 } 51 }
52 52
53 namespace internal { 53 namespace internal {
54
hlundin-webrtc 2015/11/20 12:59:22 Undo this change and you'll keep this file from th
the sun 2015/11/20 14:24:09 Done.
55 AudioSendStream::AudioSendStream( 54 AudioSendStream::AudioSendStream(
56 const webrtc::AudioSendStream::Config& config, 55 const webrtc::AudioSendStream::Config& config,
57 const rtc::scoped_refptr<webrtc::AudioState>& audio_state) 56 const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
58 : config_(config), audio_state_(audio_state) { 57 : config_(config), audio_state_(audio_state) {
59 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); 58 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
60 RTC_DCHECK_NE(config_.voe_channel_id, -1); 59 RTC_DCHECK_NE(config_.voe_channel_id, -1);
61 RTC_DCHECK(audio_state_.get()); 60 RTC_DCHECK(audio_state_.get());
62 61
63 const int channel_id = config.voe_channel_id; 62 const int channel_id = config.voe_channel_id;
64 ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine()); 63 ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
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207 206
208 VoiceEngine* AudioSendStream::voice_engine() const { 207 VoiceEngine* AudioSendStream::voice_engine() const {
209 internal::AudioState* audio_state = 208 internal::AudioState* audio_state =
210 static_cast<internal::AudioState*>(audio_state_.get()); 209 static_cast<internal::AudioState*>(audio_state_.get());
211 VoiceEngine* voice_engine = audio_state->voice_engine(); 210 VoiceEngine* voice_engine = audio_state->voice_engine();
212 RTC_DCHECK(voice_engine); 211 RTC_DCHECK(voice_engine);
213 return voice_engine; 212 return voice_engine;
214 } 213 }
215 } // namespace internal 214 } // namespace internal
216 } // namespace webrtc 215 } // namespace webrtc
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