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(2015)

Side by Side Diff: webrtc/audio/audio_send_stream.h

Issue 2844803003: Have AudioSendStream register CNG payload types with the RtpRtcpModule. (Closed)
Patch Set: Add mocks for GetRtpRtcp in AudioSendStreamTest. Created 3 years, 7 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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86 static bool ReconfigureSendCodec(AudioSendStream* stream, 86 static bool ReconfigureSendCodec(AudioSendStream* stream,
87 const Config& new_config); 87 const Config& new_config);
88 static void ReconfigureANA(AudioSendStream* stream, const Config& new_config); 88 static void ReconfigureANA(AudioSendStream* stream, const Config& new_config);
89 static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config); 89 static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config);
90 static void ReconfigureBitrateObserver(AudioSendStream* stream, 90 static void ReconfigureBitrateObserver(AudioSendStream* stream,
91 const Config& new_config); 91 const Config& new_config);
92 92
93 void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps); 93 void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps);
94 void RemoveBitrateObserver(); 94 void RemoveBitrateObserver();
95 95
96 void RegisterCngPayloadType(int payload_type, int clockrate_hz);
97
96 rtc::ThreadChecker worker_thread_checker_; 98 rtc::ThreadChecker worker_thread_checker_;
97 rtc::ThreadChecker pacer_thread_checker_; 99 rtc::ThreadChecker pacer_thread_checker_;
98 rtc::TaskQueue* worker_queue_; 100 rtc::TaskQueue* worker_queue_;
99 webrtc::AudioSendStream::Config config_; 101 webrtc::AudioSendStream::Config config_;
100 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 102 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
101 std::unique_ptr<voe::ChannelProxy> channel_proxy_; 103 std::unique_ptr<voe::ChannelProxy> channel_proxy_;
102 RtcEventLog* const event_log_; 104 RtcEventLog* const event_log_;
103 105
104 BitrateAllocator* const bitrate_allocator_; 106 BitrateAllocator* const bitrate_allocator_;
105 RtpTransportControllerSendInterface* const transport_; 107 RtpTransportControllerSendInterface* const transport_;
106 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; 108 std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
107 109
108 rtc::CriticalSection packet_loss_tracker_cs_; 110 rtc::CriticalSection packet_loss_tracker_cs_;
109 TransportFeedbackPacketLossTracker packet_loss_tracker_ 111 TransportFeedbackPacketLossTracker packet_loss_tracker_
110 GUARDED_BY(&packet_loss_tracker_cs_); 112 GUARDED_BY(&packet_loss_tracker_cs_);
111 113
112 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); 114 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
113 }; 115 };
114 } // namespace internal 116 } // namespace internal
115 } // namespace webrtc 117 } // namespace webrtc
116 118
117 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ 119 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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