Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index 020d74d82636fd4c02e868b2f750ba52076f3437..210313896004d2efd39484b2aaa52a43138b94d1 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -38,7 +38,6 @@ |
#include "webrtc/media/engine/webrtcmediaengine.h" |
#include "webrtc/media/engine/webrtcvoe.h" |
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
-#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
#include "webrtc/system_wrappers/include/field_trial.h" |
#include "webrtc/system_wrappers/include/metrics.h" |
@@ -229,8 +228,7 @@ |
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
VoEWrapper* voe_wrapper) |
- : low_priority_worker_queue_("rtc-low-prio", rtc::TaskQueue::Priority::LOW), |
- adm_(adm), |
+ : adm_(adm), |
encoder_factory_(encoder_factory), |
decoder_factory_(decoder_factory), |
voe_wrapper_(voe_wrapper) { |
@@ -689,28 +687,46 @@ |
bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, |
int64_t max_size_bytes) { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
- auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes, |
- &low_priority_worker_queue_); |
- if (!aec_dump) { |
- return false; |
- } |
- apm()->AttachAecDump(std::move(aec_dump)); |
+ FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); |
+ if (!aec_dump_file_stream) { |
+ LOG(LS_ERROR) << "Could not open AEC dump file stream."; |
+ if (!rtc::ClosePlatformFile(file)) |
+ LOG(LS_WARNING) << "Could not close file."; |
+ return false; |
+ } |
+ StopAecDump(); |
+ if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) != |
+ webrtc::AudioProcessing::kNoError) { |
+ LOG_RTCERR0(StartDebugRecording); |
+ fclose(aec_dump_file_stream); |
+ return false; |
+ } |
+ is_dumping_aec_ = true; |
return true; |
} |
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
- |
- auto aec_dump = |
- webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_); |
- if (aec_dump) { |
- apm()->AttachAecDump(std::move(aec_dump)); |
+ if (!is_dumping_aec_) { |
+ // Start dumping AEC when we are not dumping. |
+ if (apm()->StartDebugRecording(filename.c_str(), -1) != |
+ webrtc::AudioProcessing::kNoError) { |
+ LOG_RTCERR1(StartDebugRecording, filename.c_str()); |
+ } else { |
+ is_dumping_aec_ = true; |
+ } |
} |
} |
void WebRtcVoiceEngine::StopAecDump() { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
- apm()->DetachAecDump(); |
+ if (is_dumping_aec_) { |
+ // Stop dumping AEC when we are dumping. |
+ if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) { |
+ LOG_RTCERR0(StopDebugRecording); |
+ } |
+ is_dumping_aec_ = false; |
+ } |
} |
int WebRtcVoiceEngine::CreateVoEChannel() { |