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Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Issue 2986793002: Remove deprecated RtpRtcp::SetAudioPacketSize (Closed)
Patch Set: Created 3 years, 4 months ago
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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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