webrtc/voice_engine/channel_proxy.h - Issue 2964593002: Adding stats that can be used to compute output audio levels.

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Issue 2964593002: Adding stats that can be used to compute output audio levels. (Closed)

Patch Set: Add test coverage in AudioSendStreamTest. Created 3 years, 5 months ago

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1 /*	1 /*

2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.	2 * Copyright (c) 2015 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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76 virtual void RegisterReceiverCongestionControlObjects(	76 virtual void RegisterReceiverCongestionControlObjects(

77 PacketRouter* packet_router);	77 PacketRouter* packet_router);

78 virtual void ResetSenderCongestionControlObjects();	78 virtual void ResetSenderCongestionControlObjects();

79 virtual void ResetReceiverCongestionControlObjects();	79 virtual void ResetReceiverCongestionControlObjects();

80 virtual CallStatistics GetRTCPStatistics() const;	80 virtual CallStatistics GetRTCPStatistics() const;

81 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;	81 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;

82 virtual NetworkStatistics GetNetworkStatistics() const;	82 virtual NetworkStatistics GetNetworkStatistics() const;

83 virtual AudioDecodingCallStats GetDecodingCallStatistics() const;	83 virtual AudioDecodingCallStats GetDecodingCallStatistics() const;

84 virtual int GetSpeechOutputLevel() const;	84 virtual int GetSpeechOutputLevel() const;

85 virtual int GetSpeechOutputLevelFullRange() const;	85 virtual int GetSpeechOutputLevelFullRange() const;

	86 // See description of "totalAudioEnergy" in the WebRTC stats spec:

	87 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudio energy

	88 virtual double GetTotalOutputEnergy() const;

	89 virtual double GetTotalOutputDuration() const;

86 virtual uint32_t GetDelayEstimate() const;	90 virtual uint32_t GetDelayEstimate() const;

87 virtual bool SetSendTelephoneEventPayloadType(int payload_type,	91 virtual bool SetSendTelephoneEventPayloadType(int payload_type,

88 int payload_frequency);	92 int payload_frequency);

89 virtual bool SendTelephoneEventOutband(int event, int duration_ms);	93 virtual bool SendTelephoneEventOutband(int event, int duration_ms);

90 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);	94 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);

91 virtual void SetRecPayloadType(int payload_type,	95 virtual void SetRecPayloadType(int payload_type,

92 const SdpAudioFormat& format);	96 const SdpAudioFormat& format);

93 virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);	97 virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);

94 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);	98 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);

95 virtual void SetInputMute(bool muted);	99 virtual void SetInputMute(bool muted);

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138 rtc::RaceChecker audio_thread_race_checker_;	142 rtc::RaceChecker audio_thread_race_checker_;

139 rtc::RaceChecker video_capture_thread_race_checker_;	143 rtc::RaceChecker video_capture_thread_race_checker_;

140 ChannelOwner channel_owner_;	144 ChannelOwner channel_owner_;

141	145

142 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);	146 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);

143 };	147 };

144 } // namespace voe	148 } // namespace voe

145 } // namespace webrtc	149 } // namespace webrtc

146	150

147 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_	151 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_

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