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Side by Side Diff: webrtc/modules/audio_coding/main/interface/audio_coding_module.h

Issue 1362943004: ACM: Removing runtime APIs related to playout mode (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years, 2 months ago
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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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