| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index 8fcda89962ee768153b9d6337a862309b7e67aab..2ff3ad9956589d507c8c09e8e748cbc40fea3c0e 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -39,6 +39,7 @@
|
| #include "webrtc/media/engine/webrtcvoe.h"
|
| #include "webrtc/modules/audio_coding/acm2/rent_a_codec.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"
|
| @@ -565,7 +566,10 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
|
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
|
| rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
|
| VoEWrapper* voe_wrapper)
|
| - : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
|
| + : worker_queue_("file_writer_task_queue_"),
|
| + adm_(adm),
|
| + decoder_factory_(decoder_factory),
|
| + voe_wrapper_(voe_wrapper) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
|
| RTC_DCHECK(voe_wrapper);
|
| @@ -999,12 +1003,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile 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;
|
| - }
|
| + apm()->StartDebugRecording(webrtc::AecDumpFactory::Create(
|
| + aec_dump_file_stream, max_size_bytes, &worker_queue_));
|
| is_dumping_aec_ = true;
|
| return true;
|
| }
|
| @@ -1013,12 +1013,9 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| 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;
|
| - }
|
| + apm()->StartDebugRecording(
|
| + webrtc::AecDumpFactory::Create(filename.c_str(), -1, &worker_queue_));
|
| + is_dumping_aec_ = true;
|
| }
|
| }
|
|
|
| @@ -1026,9 +1023,7 @@ void WebRtcVoiceEngine::StopAecDump() {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| if (is_dumping_aec_) {
|
| // Stop dumping AEC when we are dumping.
|
| - if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) {
|
| - LOG_RTCERR0(StopDebugRecording);
|
| - }
|
| + apm()->StopDebugRecording();
|
| is_dumping_aec_ = false;
|
| }
|
| }
|
|
|