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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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