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Side by Side Diff: webrtc/voice_engine/channel_proxy.h

Issue 2516993002: Pass SdpAudioFormat through Channel, without converting to CodecInst (Closed)
Patch Set: . Created 4 years 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
(...skipping 56 matching lines...) Expand 10 before | Expand all | Expand 10 after
67 virtual CallStatistics GetRTCPStatistics() const; 67 virtual CallStatistics GetRTCPStatistics() const;
68 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; 68 virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
69 virtual NetworkStatistics GetNetworkStatistics() const; 69 virtual NetworkStatistics GetNetworkStatistics() const;
70 virtual AudioDecodingCallStats GetDecodingCallStatistics() const; 70 virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
71 virtual int32_t GetSpeechOutputLevelFullRange() const; 71 virtual int32_t GetSpeechOutputLevelFullRange() const;
72 virtual uint32_t GetDelayEstimate() const; 72 virtual uint32_t GetDelayEstimate() const;
73 virtual bool SetSendTelephoneEventPayloadType(int payload_type, 73 virtual bool SetSendTelephoneEventPayloadType(int payload_type,
74 int payload_frequency); 74 int payload_frequency);
75 virtual bool SendTelephoneEventOutband(int event, int duration_ms); 75 virtual bool SendTelephoneEventOutband(int event, int duration_ms);
76 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms); 76 virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
77 virtual int SetRecPayloadType(int payload_type, const SdpAudioFormat& format);
the sun 2016/12/08 10:53:53 remove return code since you're not checking it in
kwiberg-webrtc 2016/12/09 02:39:02 Done.
77 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink); 78 virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
78 virtual void SetInputMute(bool muted); 79 virtual void SetInputMute(bool muted);
79 virtual void RegisterExternalTransport(Transport* transport); 80 virtual void RegisterExternalTransport(Transport* transport);
80 virtual void DeRegisterExternalTransport(); 81 virtual void DeRegisterExternalTransport();
81 virtual bool ReceivedRTPPacket(const uint8_t* packet, 82 virtual bool ReceivedRTPPacket(const uint8_t* packet,
82 size_t length, 83 size_t length,
83 const PacketTime& packet_time); 84 const PacketTime& packet_time);
84 virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length); 85 virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
85 virtual const rtc::scoped_refptr<AudioDecoderFactory>& 86 virtual const rtc::scoped_refptr<AudioDecoderFactory>&
86 GetAudioDecoderFactory() const; 87 GetAudioDecoderFactory() const;
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106 rtc::ThreadChecker thread_checker_; 107 rtc::ThreadChecker thread_checker_;
107 rtc::RaceChecker race_checker_; 108 rtc::RaceChecker race_checker_;
108 ChannelOwner channel_owner_; 109 ChannelOwner channel_owner_;
109 110
110 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy); 111 RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
111 }; 112 };
112 } // namespace voe 113 } // namespace voe
113 } // namespace webrtc 114 } // namespace webrtc
114 115
115 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_ 116 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
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