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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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713 // 0 if the function succeeds. | 713 // 0 if the function succeeds. |
714 // | 714 // |
715 virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz, | 715 virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz, |
716 AudioFrame* audio_frame) = 0; | 716 AudioFrame* audio_frame) = 0; |
717 | 717 |
718 /////////////////////////////////////////////////////////////////////////// | 718 /////////////////////////////////////////////////////////////////////////// |
719 // Codec specific | 719 // Codec specific |
720 // | 720 // |
721 | 721 |
722 /////////////////////////////////////////////////////////////////////////// | 722 /////////////////////////////////////////////////////////////////////////// |
723 // int32_t SetISACMaxRate() | |
724 // Set the maximum instantaneous rate of iSAC. For a payload of B bits | |
725 // with a frame-size of T sec the instantaneous rate is B/T bits per | |
726 // second. Therefore, (B/T < |max_rate_bps|) and | |
727 // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads, | |
728 // c.f SetISACMaxPayloadSize(). | |
729 // | |
730 // Input: | |
731 // -max_rate_bps : maximum instantaneous bit-rate given in bits/sec. | |
732 // | |
733 // Return value: | |
734 // -1 if failed to set the maximum rate. | |
735 // 0 if the maximum rate is set successfully. | |
736 // | |
737 virtual int SetISACMaxRate(int max_rate_bps) = 0; | |
738 | |
739 /////////////////////////////////////////////////////////////////////////// | |
740 // int32_t SetISACMaxPayloadSize() | |
741 // Set the maximum payload size of iSAC packets. No iSAC payload, | |
742 // regardless of its frame-size, may exceed the given limit. For | |
743 // an iSAC payload of size B bits and frame-size T seconds we have; | |
744 // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f. | |
745 // SetISACMaxRate(). | |
746 // | |
747 // Input: | |
748 // -max_payload_len_bytes : maximum payload size in bytes. | |
749 // | |
750 // Return value: | |
751 // -1 if failed to set the maximum payload-size. | |
752 // 0 if the given length is set successfully. | |
753 // | |
754 virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0; | |
755 | |
756 /////////////////////////////////////////////////////////////////////////// | |
757 // int SetOpusApplication() | 723 // int SetOpusApplication() |
758 // Sets the intended application if current send codec is Opus. Opus uses this | 724 // Sets the intended application if current send codec is Opus. Opus uses this |
759 // to optimize the encoding for applications like VOIP and music. Currently, | 725 // to optimize the encoding for applications like VOIP and music. Currently, |
760 // two modes are supported: kVoip and kAudio. | 726 // two modes are supported: kVoip and kAudio. |
761 // | 727 // |
762 // Input: | 728 // Input: |
763 // - application : intended application. | 729 // - application : intended application. |
764 // | 730 // |
765 // Return value: | 731 // Return value: |
766 // -1 if current send codec is not Opus or error occurred in setting the | 732 // -1 if current send codec is not Opus or error occurred in setting the |
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1042 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; | 1008 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; |
1043 | 1009 |
1044 // Returns the timing statistics for calls to Get10MsAudio. | 1010 // Returns the timing statistics for calls to Get10MsAudio. |
1045 virtual void GetDecodingCallStatistics( | 1011 virtual void GetDecodingCallStatistics( |
1046 AudioDecodingCallStats* call_stats) const = 0; | 1012 AudioDecodingCallStats* call_stats) const = 0; |
1047 }; | 1013 }; |
1048 | 1014 |
1049 } // namespace webrtc | 1015 } // namespace webrtc |
1050 | 1016 |
1051 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ | 1017 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ |
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