Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(205)

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

Issue 2747123007: Test submission of complete AEC-dump refactoring. (Closed)
Patch Set: Changed interface and build structure after reviewer comments. Created 3 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/media/BUILD.gn ('k') | webrtc/media/engine/webrtcvoiceengine.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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/base/buffer.h" 19 #include "webrtc/base/buffer.h"
20 #include "webrtc/base/constructormagic.h" 20 #include "webrtc/base/constructormagic.h"
21 #include "webrtc/base/networkroute.h" 21 #include "webrtc/base/networkroute.h"
22 #include "webrtc/base/scoped_ref_ptr.h" 22 #include "webrtc/base/scoped_ref_ptr.h"
23 #include "webrtc/base/task_queue.h"
23 #include "webrtc/base/thread_checker.h" 24 #include "webrtc/base/thread_checker.h"
24 #include "webrtc/call/audio_state.h" 25 #include "webrtc/call/audio_state.h"
25 #include "webrtc/call/call.h" 26 #include "webrtc/call/call.h"
26 #include "webrtc/config.h" 27 #include "webrtc/config.h"
27 #include "webrtc/media/base/rtputils.h" 28 #include "webrtc/media/base/rtputils.h"
28 #include "webrtc/media/engine/apm_helpers.h" 29 #include "webrtc/media/engine/apm_helpers.h"
29 #include "webrtc/media/engine/webrtccommon.h" 30 #include "webrtc/media/engine/webrtccommon.h"
30 #include "webrtc/media/engine/webrtcvoe.h" 31 #include "webrtc/media/engine/webrtcvoe.h"
31 #include "webrtc/modules/audio_processing/include/audio_processing.h" 32 #include "webrtc/modules/audio_processing/include/audio_processing.h"
32 #include "webrtc/pc/channel.h" 33 #include "webrtc/pc/channel.h"
(...skipping 75 matching lines...) Expand 10 before | Expand all | Expand 10 after
108 void Print(webrtc::TraceLevel level, const char* trace, int length) override; 109 void Print(webrtc::TraceLevel level, const char* trace, int length) override;
109 110
110 void StartAecDump(const std::string& filename); 111 void StartAecDump(const std::string& filename);
111 int CreateVoEChannel(); 112 int CreateVoEChannel();
112 webrtc::AudioDeviceModule* adm(); 113 webrtc::AudioDeviceModule* adm();
113 webrtc::AudioProcessing* apm(); 114 webrtc::AudioProcessing* apm();
114 webrtc::voe::TransmitMixer* transmit_mixer(); 115 webrtc::voe::TransmitMixer* transmit_mixer();
115 116
116 AudioCodecs CollectRecvCodecs() const; 117 AudioCodecs CollectRecvCodecs() const;
117 118
119 rtc::TaskQueue worker_queue_;
120
118 rtc::ThreadChecker signal_thread_checker_; 121 rtc::ThreadChecker signal_thread_checker_;
119 rtc::ThreadChecker worker_thread_checker_; 122 rtc::ThreadChecker worker_thread_checker_;
120 123
121 // The audio device manager. 124 // The audio device manager.
122 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; 125 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
123 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; 126 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
124 // Reference to the APM, owned by VoE. 127 // Reference to the APM, owned by VoE.
125 webrtc::AudioProcessing* apm_ = nullptr; 128 webrtc::AudioProcessing* apm_ = nullptr;
126 // Reference to the TransmitMixer, owned by VoE. 129 // Reference to the TransmitMixer, owned by VoE.
127 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr; 130 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
(...skipping 160 matching lines...) Expand 10 before | Expand all | Expand 10 after
288 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; 291 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
289 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 292 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
290 293
291 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; 294 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
292 295
293 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 296 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
294 }; 297 };
295 } // namespace cricket 298 } // namespace cricket
296 299
297 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 300 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
OLDNEW
« no previous file with comments | « webrtc/media/BUILD.gn ('k') | webrtc/media/engine/webrtcvoiceengine.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698