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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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| 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); |
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| 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_ |
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