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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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| 406 virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; | 406 virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; |
| 407 // BWE feedback packets. | 407 // BWE feedback packets. |
| 408 virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; | 408 virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; |
| 409 | 409 |
| 410 virtual void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) = 0; | 410 virtual void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) = 0; |
| 411 | 411 |
| 412 // ************************************************************************** | 412 // ************************************************************************** |
| 413 // Audio | 413 // Audio |
| 414 // ************************************************************************** | 414 // ************************************************************************** |
| 415 | 415 |
| 416 // This function is deprecated. It was previously used to determine when it | |
| 417 // was time to send a DTMF packet in silence (CNG). | |
| 418 // Returns -1 on failure else 0. | |
| 419 RTC_DEPRECATED virtual int32_t SetAudioPacketSize( | |
| 420 uint16_t packet_size_samples) = 0; | |
| 421 | |
| 422 // Sends a TelephoneEvent tone using RFC 2833 (4733). | 416 // Sends a TelephoneEvent tone using RFC 2833 (4733). |
| 423 // Returns -1 on failure else 0. | 417 // Returns -1 on failure else 0. |
| 424 virtual int32_t SendTelephoneEventOutband(uint8_t key, | 418 virtual int32_t SendTelephoneEventOutband(uint8_t key, |
| 425 uint16_t time_ms, | 419 uint16_t time_ms, |
| 426 uint8_t level) = 0; | 420 uint8_t level) = 0; |
| 427 | 421 |
| 428 // Store the audio level in dBov for header-extension-for-audio-level- | 422 // Store the audio level in dBov for header-extension-for-audio-level- |
| 429 // indication. | 423 // indication. |
| 430 // This API shall be called before transmision of an RTP packet to ensure | 424 // This API shall be called before transmision of an RTP packet to ensure |
| 431 // that the |level| part of the extended RTP header is updated. | 425 // that the |level| part of the extended RTP header is updated. |
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| 463 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 457 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
| 464 | 458 |
| 465 // Sends a request for a keyframe. | 459 // Sends a request for a keyframe. |
| 466 // Returns -1 on failure else 0. | 460 // Returns -1 on failure else 0. |
| 467 virtual int32_t RequestKeyFrame() = 0; | 461 virtual int32_t RequestKeyFrame() = 0; |
| 468 }; | 462 }; |
| 469 | 463 |
| 470 } // namespace webrtc | 464 } // namespace webrtc |
| 471 | 465 |
| 472 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 466 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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