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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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279 int SetMinimumPlayoutDelay(int delayMs); | 279 int SetMinimumPlayoutDelay(int delayMs); |
280 int GetPlayoutTimestamp(unsigned int& timestamp); | 280 int GetPlayoutTimestamp(unsigned int& timestamp); |
281 int SetInitTimestamp(unsigned int timestamp); | 281 int SetInitTimestamp(unsigned int timestamp); |
282 int SetInitSequenceNumber(short sequenceNumber); | 282 int SetInitSequenceNumber(short sequenceNumber); |
283 | 283 |
284 // VoEVideoSyncExtended | 284 // VoEVideoSyncExtended |
285 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 285 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
286 | 286 |
287 // DTMF | 287 // DTMF |
288 int SendTelephoneEventOutband(int event, int duration_ms); | 288 int SendTelephoneEventOutband(int event, int duration_ms); |
289 int SetSendTelephoneEventPayloadType(int payload_type); | 289 int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency); |
290 | 290 |
291 // VoEAudioProcessingImpl | 291 // VoEAudioProcessingImpl |
292 int VoiceActivityIndicator(int& activity); | 292 int VoiceActivityIndicator(int& activity); |
293 | 293 |
294 // VoERTP_RTCP | 294 // VoERTP_RTCP |
295 int SetLocalSSRC(unsigned int ssrc); | 295 int SetLocalSSRC(unsigned int ssrc); |
296 int GetLocalSSRC(unsigned int& ssrc); | 296 int GetLocalSSRC(unsigned int& ssrc); |
297 int GetRemoteSSRC(unsigned int& ssrc); | 297 int GetRemoteSSRC(unsigned int& ssrc); |
298 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); | 298 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); |
299 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); | 299 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id); |
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542 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 542 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
543 | 543 |
544 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 544 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
545 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 545 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
546 }; | 546 }; |
547 | 547 |
548 } // namespace voe | 548 } // namespace voe |
549 } // namespace webrtc | 549 } // namespace webrtc |
550 | 550 |
551 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 551 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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