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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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