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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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611 | 611 |
612 // | 612 // |
613 // The shortest latency, in milliseconds, required by jitter buffer. This | 613 // The shortest latency, in milliseconds, required by jitter buffer. This |
614 // is computed based on inter-arrival times and playout mode of NetEq. The | 614 // is computed based on inter-arrival times and playout mode of NetEq. The |
615 // actual delay is the maximum of least-required-delay and the minimum-delay | 615 // actual delay is the maximum of least-required-delay and the minimum-delay |
616 // specified by SetMinumumPlayoutDelay() API. | 616 // specified by SetMinumumPlayoutDelay() API. |
617 // | 617 // |
618 virtual int LeastRequiredDelayMs() const = 0; | 618 virtual int LeastRequiredDelayMs() const = 0; |
619 | 619 |
620 /////////////////////////////////////////////////////////////////////////// | 620 /////////////////////////////////////////////////////////////////////////// |
621 // int32_t SetDtmfPlayoutStatus() | |
622 // Configure DTMF playout, i.e. whether out-of-band | |
623 // DTMF tones are played or not. | |
624 // | |
625 // Input: | |
626 // -enable : if true to enable playout out-of-band DTMF tones, | |
627 // false to disable. | |
628 // | |
629 // Return value: | |
630 // -1 if the method fails, e.g. DTMF playout is not supported. | |
631 // 0 if the status is set successfully. | |
632 // | |
633 virtual int32_t SetDtmfPlayoutStatus(const bool enable) = 0; | |
634 | |
635 /////////////////////////////////////////////////////////////////////////// | |
636 // bool DtmfPlayoutStatus() | |
637 // Get Dtmf playout status. | |
638 // | |
639 // Return value: | |
640 // true if out-of-band Dtmf tones are played, | |
641 // false if playout of Dtmf tones is disabled. | |
642 // | |
643 virtual bool DtmfPlayoutStatus() const = 0; | |
644 | |
645 /////////////////////////////////////////////////////////////////////////// | |
646 // int32_t PlayoutTimestamp() | 621 // int32_t PlayoutTimestamp() |
647 // The send timestamp of an RTP packet is associated with the decoded | 622 // The send timestamp of an RTP packet is associated with the decoded |
648 // audio of the packet in question. This function returns the timestamp of | 623 // audio of the packet in question. This function returns the timestamp of |
649 // the latest audio obtained by calling PlayoutData10ms(). | 624 // the latest audio obtained by calling PlayoutData10ms(). |
650 // | 625 // |
651 // Input: | 626 // Input: |
652 // -timestamp : a reference to a uint32_t to receive the | 627 // -timestamp : a reference to a uint32_t to receive the |
653 // timestamp. | 628 // timestamp. |
654 // Return value: | 629 // Return value: |
655 // 0 if the output is a correct timestamp. | 630 // 0 if the output is a correct timestamp. |
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856 old_config.id = 0; | 831 old_config.id = 0; |
857 old_config.neteq_config = neteq_config; | 832 old_config.neteq_config = neteq_config; |
858 old_config.clock = clock; | 833 old_config.clock = clock; |
859 return old_config; | 834 return old_config; |
860 } | 835 } |
861 | 836 |
862 NetEq::Config neteq_config; | 837 NetEq::Config neteq_config; |
863 Clock* clock; | 838 Clock* clock; |
864 AudioPacketizationCallback* transport; | 839 AudioPacketizationCallback* transport; |
865 ACMVADCallback* vad_callback; | 840 ACMVADCallback* vad_callback; |
866 bool play_dtmf; | |
867 int initial_playout_delay_ms; | 841 int initial_playout_delay_ms; |
868 int playout_channels; | 842 int playout_channels; |
869 int playout_frequency_hz; | 843 int playout_frequency_hz; |
870 }; | 844 }; |
871 | 845 |
872 static AudioCoding* Create(const Config& config); | 846 static AudioCoding* Create(const Config& config); |
873 virtual ~AudioCoding() {}; | 847 virtual ~AudioCoding() {}; |
874 | 848 |
875 // Registers a codec, specified by |send_codec|, as sending codec. | 849 // Registers a codec, specified by |send_codec|, as sending codec. |
876 // This API can be called multiple times. The last codec registered overwrites | 850 // This API can be called multiple times. The last codec registered overwrites |
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1011 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; | 985 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; |
1012 | 986 |
1013 // Returns the timing statistics for calls to Get10MsAudio. | 987 // Returns the timing statistics for calls to Get10MsAudio. |
1014 virtual void GetDecodingCallStatistics( | 988 virtual void GetDecodingCallStatistics( |
1015 AudioDecodingCallStats* call_stats) const = 0; | 989 AudioDecodingCallStats* call_stats) const = 0; |
1016 }; | 990 }; |
1017 | 991 |
1018 } // namespace webrtc | 992 } // namespace webrtc |
1019 | 993 |
1020 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ | 994 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ |
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