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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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| 599 // -timestamp : a reference to a uint32_t to receive the | 599 // -timestamp : a reference to a uint32_t to receive the |
| 600 // timestamp. | 600 // timestamp. |
| 601 // Return value: | 601 // Return value: |
| 602 // 0 if the output is a correct timestamp. | 602 // 0 if the output is a correct timestamp. |
| 603 // -1 if failed to output the correct timestamp. | 603 // -1 if failed to output the correct timestamp. |
| 604 // | 604 // |
| 605 // TODO(tlegrand): Change function to return the timestamp. | 605 // TODO(tlegrand): Change function to return the timestamp. |
| 606 virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0; | 606 virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0; |
| 607 | 607 |
| 608 /////////////////////////////////////////////////////////////////////////// | 608 /////////////////////////////////////////////////////////////////////////// |
| 609 // int32_t SetPlayoutMode() | |
| 610 // Call this API to set the playout mode. Playout mode could be optimized | |
| 611 // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is | |
| 612 // optimized to deliver highest audio quality while maintaining a minimum | |
| 613 // delay. In FAX mode, NetEQ is optimized to have few delay changes as | |
| 614 // possible and maintain a constant delay, perhaps large relative to voice | |
| 615 // mode, to avoid PLC. In streaming mode, we tolerate a little more delay | |
| 616 // to achieve better jitter robustness. | |
| 617 // | |
| 618 // Input: | |
| 619 // -mode : playout mode. Possible inputs are: | |
| 620 // "voice", | |
| 621 // "fax" and | |
| 622 // "streaming". | |
| 623 // | |
| 624 // Return value: | |
| 625 // -1 if failed to set the mode, | |
| 626 // 0 if succeeding. | |
| 627 // | |
| 628 virtual int32_t SetPlayoutMode(const AudioPlayoutMode mode) = 0; | |
| 629 | |
| 630 /////////////////////////////////////////////////////////////////////////// | |
| 631 // AudioPlayoutMode PlayoutMode() | |
| 632 // Get playout mode, i.e. whether it is speech, FAX or streaming. See | |
| 633 // audio_coding_module_typedefs.h for definition of AudioPlayoutMode. | |
| 634 // | |
| 635 // Return value: | |
| 636 // voice: is for voice output, | |
| 637 // fax: a mode that is optimized for receiving FAX signals. | |
| 638 // In this mode NetEq tries to maintain a constant high | |
| 639 // delay to avoid PLC if possible. | |
| 640 // streaming: a mode that is suitable for streaming. In this mode we | |
| 641 // accept longer delay to improve jitter robustness. | |
| 642 // | |
| 643 virtual AudioPlayoutMode PlayoutMode() const = 0; | |
| 644 | |
| 645 /////////////////////////////////////////////////////////////////////////// | |
| 646 // int32_t PlayoutData10Ms( | 609 // int32_t PlayoutData10Ms( |
| 647 // Get 10 milliseconds of raw audio data for playout, at the given sampling | 610 // Get 10 milliseconds of raw audio data for playout, at the given sampling |
| 648 // frequency. ACM will perform a resampling if required. | 611 // frequency. ACM will perform a resampling if required. |
| 649 // | 612 // |
| 650 // Input: | 613 // Input: |
| 651 // -desired_freq_hz : the desired sampling frequency, in Hertz, of the | 614 // -desired_freq_hz : the desired sampling frequency, in Hertz, of the |
| 652 // output audio. If set to -1, the function returns | 615 // output audio. If set to -1, the function returns |
| 653 // the audio at the current sampling frequency. | 616 // the audio at the current sampling frequency. |
| 654 // | 617 // |
| 655 // Output: | 618 // Output: |
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| 957 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; | 920 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; |
| 958 | 921 |
| 959 // Returns the timing statistics for calls to Get10MsAudio. | 922 // Returns the timing statistics for calls to Get10MsAudio. |
| 960 virtual void GetDecodingCallStatistics( | 923 virtual void GetDecodingCallStatistics( |
| 961 AudioDecodingCallStats* call_stats) const = 0; | 924 AudioDecodingCallStats* call_stats) const = 0; |
| 962 }; | 925 }; |
| 963 | 926 |
| 964 } // namespace webrtc | 927 } // namespace webrtc |
| 965 | 928 |
| 966 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ | 929 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ |
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