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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 1803063004: Reset the BWE when the network changes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Removed changes in call dir and leave that in a separate CL. Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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
11 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 11 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
12 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 12 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory> 15 #include <memory>
16 #include <string> 16 #include <string>
17 #include <vector> 17 #include <vector>
18 18
19 #include "webrtc/audio_state.h" 19 #include "webrtc/audio_state.h"
20 #include "webrtc/base/buffer.h" 20 #include "webrtc/base/buffer.h"
21 #include "webrtc/base/networkroute.h"
21 #include "webrtc/base/stream.h" 22 #include "webrtc/base/stream.h"
22 #include "webrtc/base/thread_checker.h" 23 #include "webrtc/base/thread_checker.h"
23 #include "webrtc/call.h" 24 #include "webrtc/call.h"
24 #include "webrtc/common.h" 25 #include "webrtc/common.h"
25 #include "webrtc/config.h" 26 #include "webrtc/config.h"
26 #include "webrtc/media/base/rtputils.h" 27 #include "webrtc/media/base/rtputils.h"
27 #include "webrtc/media/engine/webrtccommon.h" 28 #include "webrtc/media/engine/webrtccommon.h"
28 #include "webrtc/media/engine/webrtcvoe.h" 29 #include "webrtc/media/engine/webrtcvoe.h"
29 #include "webrtc/pc/channel.h" 30 #include "webrtc/pc/channel.h"
30 31
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176 int type_event_delay) override; 177 int type_event_delay) override;
177 bool SetOutputVolume(uint32_t ssrc, double volume) override; 178 bool SetOutputVolume(uint32_t ssrc, double volume) override;
178 179
179 bool CanInsertDtmf() override; 180 bool CanInsertDtmf() override;
180 bool InsertDtmf(uint32_t ssrc, int event, int duration) override; 181 bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
181 182
182 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, 183 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
183 const rtc::PacketTime& packet_time) override; 184 const rtc::PacketTime& packet_time) override;
184 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, 185 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
185 const rtc::PacketTime& packet_time) override; 186 const rtc::PacketTime& packet_time) override;
187 void OnNetworkRouteChanged(const std::string& transport_name,
188 const NetworkRoute& network_route) override;
186 void OnReadyToSend(bool ready) override; 189 void OnReadyToSend(bool ready) override;
187 bool GetStats(VoiceMediaInfo* info) override; 190 bool GetStats(VoiceMediaInfo* info) override;
188 191
189 void SetRawAudioSink( 192 void SetRawAudioSink(
190 uint32_t ssrc, 193 uint32_t ssrc,
191 std::unique_ptr<webrtc::AudioSinkInterface> sink) override; 194 std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
192 195
193 // implements Transport interface 196 // implements Transport interface
194 bool SendRtp(const uint8_t* data, 197 bool SendRtp(const uint8_t* data,
195 size_t len, 198 size_t len,
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282 int cng_payload_type = -1; 285 int cng_payload_type = -1;
283 int cng_plfreq = -1; 286 int cng_plfreq = -1;
284 webrtc::CodecInst codec_inst; 287 webrtc::CodecInst codec_inst;
285 } send_codec_spec_; 288 } send_codec_spec_;
286 289
287 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 290 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
288 }; 291 };
289 } // namespace cricket 292 } // namespace cricket
290 293
291 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 294 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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