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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/api/audio_codecs/audio_encoder_factory.h" | 19 #include "webrtc/api/audio_codecs/audio_encoder_factory.h" |
20 #include "webrtc/api/rtpreceiverinterface.h" | 20 #include "webrtc/api/rtpreceiverinterface.h" |
21 #include "webrtc/base/buffer.h" | 21 #include "webrtc/base/buffer.h" |
22 #include "webrtc/base/constructormagic.h" | 22 #include "webrtc/base/constructormagic.h" |
23 #include "webrtc/base/networkroute.h" | 23 #include "webrtc/base/networkroute.h" |
24 #include "webrtc/base/scoped_ref_ptr.h" | 24 #include "webrtc/base/scoped_ref_ptr.h" |
25 #include "webrtc/base/task_queue.h" | |
26 #include "webrtc/base/thread_checker.h" | 25 #include "webrtc/base/thread_checker.h" |
27 #include "webrtc/call/audio_state.h" | 26 #include "webrtc/call/audio_state.h" |
28 #include "webrtc/call/call.h" | 27 #include "webrtc/call/call.h" |
29 #include "webrtc/config.h" | 28 #include "webrtc/config.h" |
30 #include "webrtc/media/base/rtputils.h" | 29 #include "webrtc/media/base/rtputils.h" |
31 #include "webrtc/media/engine/apm_helpers.h" | 30 #include "webrtc/media/engine/apm_helpers.h" |
32 #include "webrtc/media/engine/webrtccommon.h" | 31 #include "webrtc/media/engine/webrtccommon.h" |
33 #include "webrtc/media/engine/webrtcvoe.h" | 32 #include "webrtc/media/engine/webrtcvoe.h" |
34 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 33 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
35 #include "webrtc/pc/channel.h" | 34 #include "webrtc/pc/channel.h" |
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104 // Every option that is "set" will be applied. Every option not "set" will be | 103 // Every option that is "set" will be applied. Every option not "set" will be |
105 // ignored. This allows us to selectively turn on and off different options | 104 // ignored. This allows us to selectively turn on and off different options |
106 // easily at any time. | 105 // easily at any time. |
107 bool ApplyOptions(const AudioOptions& options); | 106 bool ApplyOptions(const AudioOptions& options); |
108 | 107 |
109 // webrtc::TraceCallback: | 108 // webrtc::TraceCallback: |
110 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 109 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
111 | 110 |
112 void StartAecDump(const std::string& filename); | 111 void StartAecDump(const std::string& filename); |
113 int CreateVoEChannel(); | 112 int CreateVoEChannel(); |
114 | |
115 rtc::TaskQueue low_priority_worker_queue_; | |
116 | |
117 webrtc::AudioDeviceModule* adm(); | 113 webrtc::AudioDeviceModule* adm(); |
118 webrtc::AudioProcessing* apm(); | 114 webrtc::AudioProcessing* apm(); |
119 webrtc::voe::TransmitMixer* transmit_mixer(); | 115 webrtc::voe::TransmitMixer* transmit_mixer(); |
120 | 116 |
121 AudioCodecs CollectCodecs( | 117 AudioCodecs CollectCodecs( |
122 const std::vector<webrtc::AudioCodecSpec>& specs) const; | 118 const std::vector<webrtc::AudioCodecSpec>& specs) const; |
123 | 119 |
124 rtc::ThreadChecker signal_thread_checker_; | 120 rtc::ThreadChecker signal_thread_checker_; |
125 rtc::ThreadChecker worker_thread_checker_; | 121 rtc::ThreadChecker worker_thread_checker_; |
126 | 122 |
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298 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 294 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
299 | 295 |
300 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> | 296 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> |
301 send_codec_spec_; | 297 send_codec_spec_; |
302 | 298 |
303 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 299 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
304 }; | 300 }; |
305 } // namespace cricket | 301 } // namespace cricket |
306 | 302 |
307 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 303 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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