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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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253 // (XR) VOIP metric. | 253 // (XR) VOIP metric. |
254 int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) override; | 254 int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) override; |
255 | 255 |
256 // (XR) Receiver reference time report. | 256 // (XR) Receiver reference time report. |
257 void SetRtcpXrRrtrStatus(bool enable) override; | 257 void SetRtcpXrRrtrStatus(bool enable) override; |
258 | 258 |
259 bool RtcpXrRrtrStatus() const override; | 259 bool RtcpXrRrtrStatus() const override; |
260 | 260 |
261 // Audio part. | 261 // Audio part. |
262 | 262 |
263 // Set audio packet size, used to determine when it's time to send a DTMF | 263 // This function is deprecated. It was previously used to determine when it |
264 // packet in silence (CNG). | 264 // was time to send a DTMF packet in silence (CNG). |
265 int32_t SetAudioPacketSize(uint16_t packet_size_samples) override; | 265 int32_t SetAudioPacketSize(uint16_t packet_size_samples) override; |
266 | 266 |
267 // Send a TelephoneEvent tone using RFC 2833 (4733). | 267 // Send a TelephoneEvent tone using RFC 2833 (4733). |
268 int32_t SendTelephoneEventOutband(uint8_t key, | 268 int32_t SendTelephoneEventOutband(uint8_t key, |
269 uint16_t time_ms, | 269 uint16_t time_ms, |
270 uint8_t level) override; | 270 uint8_t level) override; |
271 | 271 |
272 // Store the audio level in d_bov for header-extension-for-audio-level- | 272 // Store the audio level in d_bov for header-extension-for-audio-level- |
273 // indication. | 273 // indication. |
274 int32_t SetAudioLevel(uint8_t level_d_bov) override; | 274 int32_t SetAudioLevel(uint8_t level_d_bov) override; |
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356 PacketLossStats receive_loss_stats_; | 356 PacketLossStats receive_loss_stats_; |
357 | 357 |
358 // The processed RTT from RtcpRttStats. | 358 // The processed RTT from RtcpRttStats. |
359 rtc::CriticalSection critical_section_rtt_; | 359 rtc::CriticalSection critical_section_rtt_; |
360 int64_t rtt_ms_; | 360 int64_t rtt_ms_; |
361 }; | 361 }; |
362 | 362 |
363 } // namespace webrtc | 363 } // namespace webrtc |
364 | 364 |
365 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ | 365 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |
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