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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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435 // packet in silence (CNG). | 435 // packet in silence (CNG). |
436 // Returns -1 on failure else 0. | 436 // Returns -1 on failure else 0. |
437 virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; | 437 virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; |
438 | 438 |
439 // Sends a TelephoneEvent tone using RFC 2833 (4733). | 439 // Sends a TelephoneEvent tone using RFC 2833 (4733). |
440 // Returns -1 on failure else 0. | 440 // Returns -1 on failure else 0. |
441 virtual int32_t SendTelephoneEventOutband(uint8_t key, | 441 virtual int32_t SendTelephoneEventOutband(uint8_t key, |
442 uint16_t time_ms, | 442 uint16_t time_ms, |
443 uint8_t level) = 0; | 443 uint8_t level) = 0; |
444 | 444 |
445 // Sets payload type for Redundant Audio Data RFC 2198. | |
446 // Returns -1 on failure else 0. | |
447 virtual int32_t SetSendREDPayloadType(int8_t payload_type) = 0; | |
448 | |
449 // Get payload type for Redundant Audio Data RFC 2198. | |
450 // Returns -1 on failure else 0. | |
451 virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 0; | |
452 | |
453 // Store the audio level in dBov for header-extension-for-audio-level- | 445 // Store the audio level in dBov for header-extension-for-audio-level- |
454 // indication. | 446 // indication. |
455 // This API shall be called before transmision of an RTP packet to ensure | 447 // This API shall be called before transmision of an RTP packet to ensure |
456 // that the |level| part of the extended RTP header is updated. | 448 // that the |level| part of the extended RTP header is updated. |
457 // return -1 on failure else 0. | 449 // return -1 on failure else 0. |
458 virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0; | 450 virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0; |
459 | 451 |
460 // ************************************************************************** | 452 // ************************************************************************** |
461 // Video | 453 // Video |
462 // ************************************************************************** | 454 // ************************************************************************** |
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479 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 471 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
480 | 472 |
481 // Sends a request for a keyframe. | 473 // Sends a request for a keyframe. |
482 // Returns -1 on failure else 0. | 474 // Returns -1 on failure else 0. |
483 virtual int32_t RequestKeyFrame() = 0; | 475 virtual int32_t RequestKeyFrame() = 0; |
484 }; | 476 }; |
485 | 477 |
486 } // namespace webrtc | 478 } // namespace webrtc |
487 | 479 |
488 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 480 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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