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Issue 1718473002: Move congestion controller to a separate module. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add presubmit Created 4 years, 10 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
11 #include "webrtc/audio/audio_receive_stream.h" 11 #include "webrtc/audio/audio_receive_stream.h"
12 12
13 #include <string> 13 #include <string>
14 #include <utility> 14 #include <utility>
15 15
16 #include "webrtc/audio/audio_sink.h" 16 #include "webrtc/audio/audio_sink.h"
17 #include "webrtc/audio/audio_state.h" 17 #include "webrtc/audio/audio_state.h"
18 #include "webrtc/audio/conversion.h" 18 #include "webrtc/audio/conversion.h"
19 #include "webrtc/base/checks.h" 19 #include "webrtc/base/checks.h"
20 #include "webrtc/base/logging.h" 20 #include "webrtc/base/logging.h"
21 #include "webrtc/call/congestion_controller.h" 21 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
22 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" 22 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h"
23 #include "webrtc/system_wrappers/include/tick_util.h" 23 #include "webrtc/system_wrappers/include/tick_util.h"
24 #include "webrtc/voice_engine/channel_proxy.h" 24 #include "webrtc/voice_engine/channel_proxy.h"
25 #include "webrtc/voice_engine/include/voe_base.h" 25 #include "webrtc/voice_engine/include/voe_base.h"
26 #include "webrtc/voice_engine/include/voe_codec.h" 26 #include "webrtc/voice_engine/include/voe_codec.h"
27 #include "webrtc/voice_engine/include/voe_neteq_stats.h" 27 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
28 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 28 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
29 #include "webrtc/voice_engine/include/voe_video_sync.h" 29 #include "webrtc/voice_engine/include/voe_video_sync.h"
30 #include "webrtc/voice_engine/include/voe_volume_control.h" 30 #include "webrtc/voice_engine/include/voe_volume_control.h"
31 #include "webrtc/voice_engine/voice_engine_impl.h" 31 #include "webrtc/voice_engine/voice_engine_impl.h"
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241 241
242 VoiceEngine* AudioReceiveStream::voice_engine() const { 242 VoiceEngine* AudioReceiveStream::voice_engine() const {
243 internal::AudioState* audio_state = 243 internal::AudioState* audio_state =
244 static_cast<internal::AudioState*>(audio_state_.get()); 244 static_cast<internal::AudioState*>(audio_state_.get());
245 VoiceEngine* voice_engine = audio_state->voice_engine(); 245 VoiceEngine* voice_engine = audio_state->voice_engine();
246 RTC_DCHECK(voice_engine); 246 RTC_DCHECK(voice_engine);
247 return voice_engine; 247 return voice_engine;
248 } 248 }
249 } // namespace internal 249 } // namespace internal
250 } // namespace webrtc 250 } // namespace webrtc
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