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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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399 // < 0, on error. | 399 // < 0, on error. |
400 virtual int32_t SetRenderDelay(uint32_t timeMS) = 0; | 400 virtual int32_t SetRenderDelay(uint32_t timeMS) = 0; |
401 | 401 |
402 // The total delay desired by the VCM. Can be less than the minimum | 402 // The total delay desired by the VCM. Can be less than the minimum |
403 // delay set with SetMinimumPlayoutDelay. | 403 // delay set with SetMinimumPlayoutDelay. |
404 // | 404 // |
405 // Return value : Total delay in ms, on success. | 405 // Return value : Total delay in ms, on success. |
406 // < 0, on error. | 406 // < 0, on error. |
407 virtual int32_t Delay() const = 0; | 407 virtual int32_t Delay() const = 0; |
408 | 408 |
409 // Returns the number of packets discarded by the jitter buffer due to being | |
410 // too late. This can include duplicated packets which arrived after the | |
411 // frame was sent to the decoder. Therefore packets which were prematurely | |
412 // NACKed will be counted. | |
413 virtual uint32_t DiscardedPackets() const = 0; | |
414 | |
415 // Robustness APIs | 409 // Robustness APIs |
416 | 410 |
417 // Set the receiver robustness mode. The mode decides how the receiver | 411 // Set the receiver robustness mode. The mode decides how the receiver |
418 // responds to losses in the stream. The type of counter-measure (soft or | 412 // responds to losses in the stream. The type of counter-measure (soft or |
419 // hard NACK, dual decoder, RPS, etc.) is selected through the | 413 // hard NACK, dual decoder, RPS, etc.) is selected through the |
420 // robustnessMode parameter. The errorMode parameter decides if it is | 414 // robustnessMode parameter. The errorMode parameter decides if it is |
421 // allowed to display frames corrupted by losses. Note that not all | 415 // allowed to display frames corrupted by losses. Note that not all |
422 // combinations of the two parameters are feasible. An error will be | 416 // combinations of the two parameters are feasible. An error will be |
423 // returned for invalid combinations. | 417 // returned for invalid combinations. |
424 // Input: | 418 // Input: |
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453 | 447 |
454 virtual void RegisterPostEncodeImageCallback( | 448 virtual void RegisterPostEncodeImageCallback( |
455 EncodedImageCallback* post_encode_callback) = 0; | 449 EncodedImageCallback* post_encode_callback) = 0; |
456 // Releases pending decode calls, permitting faster thread shutdown. | 450 // Releases pending decode calls, permitting faster thread shutdown. |
457 virtual void TriggerDecoderShutdown() = 0; | 451 virtual void TriggerDecoderShutdown() = 0; |
458 }; | 452 }; |
459 | 453 |
460 } // namespace webrtc | 454 } // namespace webrtc |
461 | 455 |
462 #endif // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_ | 456 #endif // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_ |
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