OLD | NEW |
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 |
(...skipping 275 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
286 int SetMinimumPlayoutDelay(int delayMs); | 286 int SetMinimumPlayoutDelay(int delayMs); |
287 int GetPlayoutTimestamp(unsigned int& timestamp); | 287 int GetPlayoutTimestamp(unsigned int& timestamp); |
288 int SetInitTimestamp(unsigned int timestamp); | 288 int SetInitTimestamp(unsigned int timestamp); |
289 int SetInitSequenceNumber(short sequenceNumber); | 289 int SetInitSequenceNumber(short sequenceNumber); |
290 | 290 |
291 // VoEVideoSyncExtended | 291 // VoEVideoSyncExtended |
292 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 292 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
293 | 293 |
294 // DTMF | 294 // DTMF |
295 int SendTelephoneEventOutband(int event, int duration_ms); | 295 int SendTelephoneEventOutband(int event, int duration_ms); |
296 int SetSendTelephoneEventPayloadType(int payload_type); | 296 int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency); |
297 | 297 |
298 // VoEAudioProcessingImpl | 298 // VoEAudioProcessingImpl |
299 int VoiceActivityIndicator(int& activity); | 299 int VoiceActivityIndicator(int& activity); |
300 | 300 |
301 // VoERTP_RTCP | 301 // VoERTP_RTCP |
302 int SetLocalSSRC(unsigned int ssrc); | 302 int SetLocalSSRC(unsigned int ssrc); |
303 int GetLocalSSRC(unsigned int& ssrc); | 303 int GetLocalSSRC(unsigned int& ssrc); |
304 int GetRemoteSSRC(unsigned int& ssrc); | 304 int GetRemoteSSRC(unsigned int& ssrc); |
305 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); | 305 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); |
306 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); | 306 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); |
(...skipping 241 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
548 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 548 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
549 | 549 |
550 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 550 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
551 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 551 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
552 }; | 552 }; |
553 | 553 |
554 } // namespace voe | 554 } // namespace voe |
555 } // namespace webrtc | 555 } // namespace webrtc |
556 | 556 |
557 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 557 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
OLD | NEW |