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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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421 virtual void RegisterRtcpStatisticsCallback( | 421 virtual void RegisterRtcpStatisticsCallback( |
422 RtcpStatisticsCallback* callback) = 0; | 422 RtcpStatisticsCallback* callback) = 0; |
423 virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; | 423 virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; |
424 // BWE feedback packets. | 424 // BWE feedback packets. |
425 virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; | 425 virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; |
426 | 426 |
427 // ************************************************************************** | 427 // ************************************************************************** |
428 // Audio | 428 // Audio |
429 // ************************************************************************** | 429 // ************************************************************************** |
430 | 430 |
431 // Sets audio packet size, used to determine when it's time to send a DTMF | 431 // This function is deprecated. It was previously used to determine when it |
432 // packet in silence (CNG). | 432 // was time to send a DTMF packet in silence (CNG). |
433 // Returns -1 on failure else 0. | 433 // Returns -1 on failure else 0. |
434 virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; | 434 RTC_DEPRECATED virtual int32_t SetAudioPacketSize( |
| 435 uint16_t packet_size_samples) = 0; |
435 | 436 |
436 // Sends a TelephoneEvent tone using RFC 2833 (4733). | 437 // Sends a TelephoneEvent tone using RFC 2833 (4733). |
437 // Returns -1 on failure else 0. | 438 // Returns -1 on failure else 0. |
438 virtual int32_t SendTelephoneEventOutband(uint8_t key, | 439 virtual int32_t SendTelephoneEventOutband(uint8_t key, |
439 uint16_t time_ms, | 440 uint16_t time_ms, |
440 uint8_t level) = 0; | 441 uint8_t level) = 0; |
441 | 442 |
442 // Store the audio level in dBov for header-extension-for-audio-level- | 443 // Store the audio level in dBov for header-extension-for-audio-level- |
443 // indication. | 444 // indication. |
444 // This API shall be called before transmision of an RTP packet to ensure | 445 // This API shall be called before transmision of an RTP packet to ensure |
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477 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 478 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
478 | 479 |
479 // Sends a request for a keyframe. | 480 // Sends a request for a keyframe. |
480 // Returns -1 on failure else 0. | 481 // Returns -1 on failure else 0. |
481 virtual int32_t RequestKeyFrame() = 0; | 482 virtual int32_t RequestKeyFrame() = 0; |
482 }; | 483 }; |
483 | 484 |
484 } // namespace webrtc | 485 } // namespace webrtc |
485 | 486 |
486 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 487 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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