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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 |
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31 #include "webrtc/base/trace_event.h" | 31 #include "webrtc/base/trace_event.h" |
32 #include "webrtc/media/base/audiosource.h" | 32 #include "webrtc/media/base/audiosource.h" |
33 #include "webrtc/media/base/mediaconstants.h" | 33 #include "webrtc/media/base/mediaconstants.h" |
34 #include "webrtc/media/base/streamparams.h" | 34 #include "webrtc/media/base/streamparams.h" |
35 #include "webrtc/media/engine/adm_helpers.h" | 35 #include "webrtc/media/engine/adm_helpers.h" |
36 #include "webrtc/media/engine/apm_helpers.h" | 36 #include "webrtc/media/engine/apm_helpers.h" |
37 #include "webrtc/media/engine/payload_type_mapper.h" | 37 #include "webrtc/media/engine/payload_type_mapper.h" |
38 #include "webrtc/media/engine/webrtcmediaengine.h" | 38 #include "webrtc/media/engine/webrtcmediaengine.h" |
39 #include "webrtc/media/engine/webrtcvoe.h" | 39 #include "webrtc/media/engine/webrtcvoe.h" |
40 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 40 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
41 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" | |
42 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 41 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
43 #include "webrtc/system_wrappers/include/field_trial.h" | 42 #include "webrtc/system_wrappers/include/field_trial.h" |
44 #include "webrtc/system_wrappers/include/metrics.h" | 43 #include "webrtc/system_wrappers/include/metrics.h" |
45 #include "webrtc/system_wrappers/include/trace.h" | 44 #include "webrtc/system_wrappers/include/trace.h" |
46 #include "webrtc/voice_engine/transmit_mixer.h" | 45 #include "webrtc/voice_engine/transmit_mixer.h" |
47 | 46 |
48 namespace cricket { | 47 namespace cricket { |
49 namespace { | 48 namespace { |
50 | 49 |
51 constexpr size_t kMaxUnsignaledRecvStreams = 1; | 50 constexpr size_t kMaxUnsignaledRecvStreams = 1; |
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222 audio_state_ = | 221 audio_state_ = |
223 webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer)); | 222 webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer)); |
224 } | 223 } |
225 | 224 |
226 WebRtcVoiceEngine::WebRtcVoiceEngine( | 225 WebRtcVoiceEngine::WebRtcVoiceEngine( |
227 webrtc::AudioDeviceModule* adm, | 226 webrtc::AudioDeviceModule* adm, |
228 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, | 227 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
229 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, | 228 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
230 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, | 229 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
231 VoEWrapper* voe_wrapper) | 230 VoEWrapper* voe_wrapper) |
232 : low_priority_worker_queue_("rtc-low-prio", rtc::TaskQueue::Priority::LOW), | 231 : adm_(adm), |
233 adm_(adm), | |
234 encoder_factory_(encoder_factory), | 232 encoder_factory_(encoder_factory), |
235 decoder_factory_(decoder_factory), | 233 decoder_factory_(decoder_factory), |
236 voe_wrapper_(voe_wrapper) { | 234 voe_wrapper_(voe_wrapper) { |
237 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 235 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
238 LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; | 236 LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; |
239 RTC_DCHECK(voe_wrapper); | 237 RTC_DCHECK(voe_wrapper); |
240 RTC_DCHECK(decoder_factory); | 238 RTC_DCHECK(decoder_factory); |
241 | 239 |
242 signal_thread_checker_.DetachFromThread(); | 240 signal_thread_checker_.DetachFromThread(); |
243 | 241 |
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682 void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) { | 680 void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) { |
683 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 681 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
684 auto it = std::find(channels_.begin(), channels_.end(), channel); | 682 auto it = std::find(channels_.begin(), channels_.end(), channel); |
685 RTC_DCHECK(it != channels_.end()); | 683 RTC_DCHECK(it != channels_.end()); |
686 channels_.erase(it); | 684 channels_.erase(it); |
687 } | 685 } |
688 | 686 |
689 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, | 687 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, |
690 int64_t max_size_bytes) { | 688 int64_t max_size_bytes) { |
691 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 689 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
692 auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes, | 690 FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); |
693 &low_priority_worker_queue_); | 691 if (!aec_dump_file_stream) { |
694 if (!aec_dump) { | 692 LOG(LS_ERROR) << "Could not open AEC dump file stream."; |
| 693 if (!rtc::ClosePlatformFile(file)) |
| 694 LOG(LS_WARNING) << "Could not close file."; |
695 return false; | 695 return false; |
696 } | 696 } |
697 apm()->AttachAecDump(std::move(aec_dump)); | 697 StopAecDump(); |
| 698 if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) != |
| 699 webrtc::AudioProcessing::kNoError) { |
| 700 LOG_RTCERR0(StartDebugRecording); |
| 701 fclose(aec_dump_file_stream); |
| 702 return false; |
| 703 } |
| 704 is_dumping_aec_ = true; |
698 return true; | 705 return true; |
699 } | 706 } |
700 | 707 |
701 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { | 708 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
702 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 709 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
703 | 710 if (!is_dumping_aec_) { |
704 auto aec_dump = | 711 // Start dumping AEC when we are not dumping. |
705 webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_); | 712 if (apm()->StartDebugRecording(filename.c_str(), -1) != |
706 if (aec_dump) { | 713 webrtc::AudioProcessing::kNoError) { |
707 apm()->AttachAecDump(std::move(aec_dump)); | 714 LOG_RTCERR1(StartDebugRecording, filename.c_str()); |
| 715 } else { |
| 716 is_dumping_aec_ = true; |
| 717 } |
708 } | 718 } |
709 } | 719 } |
710 | 720 |
711 void WebRtcVoiceEngine::StopAecDump() { | 721 void WebRtcVoiceEngine::StopAecDump() { |
712 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 722 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
713 apm()->DetachAecDump(); | 723 if (is_dumping_aec_) { |
| 724 // Stop dumping AEC when we are dumping. |
| 725 if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) { |
| 726 LOG_RTCERR0(StopDebugRecording); |
| 727 } |
| 728 is_dumping_aec_ = false; |
| 729 } |
714 } | 730 } |
715 | 731 |
716 int WebRtcVoiceEngine::CreateVoEChannel() { | 732 int WebRtcVoiceEngine::CreateVoEChannel() { |
717 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 733 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
718 return voe_wrapper_->base()->CreateChannel(channel_config_); | 734 return voe_wrapper_->base()->CreateChannel(channel_config_); |
719 } | 735 } |
720 | 736 |
721 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { | 737 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { |
722 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 738 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
723 RTC_DCHECK(adm_); | 739 RTC_DCHECK(adm_); |
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2341 ssrc); | 2357 ssrc); |
2342 if (it != unsignaled_recv_ssrcs_.end()) { | 2358 if (it != unsignaled_recv_ssrcs_.end()) { |
2343 unsignaled_recv_ssrcs_.erase(it); | 2359 unsignaled_recv_ssrcs_.erase(it); |
2344 return true; | 2360 return true; |
2345 } | 2361 } |
2346 return false; | 2362 return false; |
2347 } | 2363 } |
2348 } // namespace cricket | 2364 } // namespace cricket |
2349 | 2365 |
2350 #endif // HAVE_WEBRTC_VOICE | 2366 #endif // HAVE_WEBRTC_VOICE |
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