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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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242 int StopRecordingPlayout(); | 242 int StopRecordingPlayout(); |
243 | 243 |
244 void SetMixWithMicStatus(bool mix); | 244 void SetMixWithMicStatus(bool mix); |
245 | 245 |
246 // Muting, Volume and Level. | 246 // Muting, Volume and Level. |
247 void SetInputMute(bool enable); | 247 void SetInputMute(bool enable); |
248 void SetChannelOutputVolumeScaling(float scaling); | 248 void SetChannelOutputVolumeScaling(float scaling); |
249 int GetSpeechOutputLevel() const; | 249 int GetSpeechOutputLevel() const; |
250 int GetSpeechOutputLevelFullRange() const; | 250 int GetSpeechOutputLevelFullRange() const; |
251 | 251 |
252 // VoENetEqStats | 252 // TODO(solenberg): |
hlundin-webrtc
2017/03/13 08:36:38
Todo what?
the sun
2017/03/13 08:52:07
"Remember to check if these are used" - they are.
| |
253 int GetNetworkStatistics(NetworkStatistics& stats); | 253 int GetNetworkStatistics(NetworkStatistics& stats); |
254 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; | 254 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; |
255 | 255 |
256 // Audio+Video Sync | 256 // Audio+Video Sync |
257 uint32_t GetDelayEstimate() const; | 257 uint32_t GetDelayEstimate() const; |
258 int SetMinimumPlayoutDelay(int delayMs); | 258 int SetMinimumPlayoutDelay(int delayMs); |
259 int GetPlayoutTimestamp(unsigned int& timestamp); | 259 int GetPlayoutTimestamp(unsigned int& timestamp); |
260 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 260 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
261 | 261 |
262 // DTMF | 262 // DTMF |
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503 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 503 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
504 | 504 |
505 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 505 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
506 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 506 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
507 }; | 507 }; |
508 | 508 |
509 } // namespace voe | 509 } // namespace voe |
510 } // namespace webrtc | 510 } // namespace webrtc |
511 | 511 |
512 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 512 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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