| Index: talk/media/webrtc/webrtcvoiceengine.cc
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
|
| index 642fd3d233e649ecbcb989a85391176b1761e0dc..358645d853f5e07ee5a5eb303e9e8fef9928cd6c 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.cc
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.cc
|
| @@ -369,7 +369,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine()
|
| adm_(NULL),
|
| log_filter_(SeverityToFilter(kDefaultLogSeverity)),
|
| is_dumping_aec_(false),
|
| - desired_local_monitor_enable_(false),
|
| tx_processor_ssrc_(0),
|
| rx_processor_ssrc_(0) {
|
| Construct();
|
| @@ -382,7 +381,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
|
| adm_(NULL),
|
| log_filter_(SeverityToFilter(kDefaultLogSeverity)),
|
| is_dumping_aec_(false),
|
| - desired_local_monitor_enable_(false),
|
| tx_processor_ssrc_(0),
|
| rx_processor_ssrc_(0) {
|
| Construct();
|
| @@ -572,7 +570,6 @@ void WebRtcVoiceEngine::Terminate() {
|
| StopAecDump();
|
|
|
| voe_wrapper_->base()->Terminate();
|
| - desired_local_monitor_enable_ = false;
|
| }
|
|
|
| int WebRtcVoiceEngine::GetCapabilities() {
|
| @@ -933,14 +930,8 @@ bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
|
| << ") and speaker to (id=" << out_id << ", name=" << out_name
|
| << ")";
|
|
|
| - // If we're running the local monitor, we need to stop it first.
|
| - bool ret = true;
|
| - if (!PauseLocalMonitor()) {
|
| - LOG(LS_WARNING) << "Failed to pause local monitor";
|
| - ret = false;
|
| - }
|
| -
|
| // Must also pause all audio playback and capture.
|
| + bool ret = true;
|
| for (WebRtcVoiceMediaChannel* channel : channels_) {
|
| if (!channel->PausePlayout()) {
|
| LOG(LS_WARNING) << "Failed to pause playout";
|
| @@ -990,12 +981,6 @@ bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
|
| }
|
| }
|
|
|
| - // Resume local monitor.
|
| - if (!ResumeLocalMonitor()) {
|
| - LOG(LS_WARNING) << "Failed to resume local monitor";
|
| - ret = false;
|
| - }
|
| -
|
| if (ret) {
|
| LOG(LS_INFO) << "Set microphone to (id=" << in_id <<" name=" << in_name
|
| << ") and speaker to (id="<< out_id << " name=" << out_name
|
| @@ -1083,42 +1068,6 @@ int WebRtcVoiceEngine::GetInputLevel() {
|
| static_cast<int>(ulevel) : -1;
|
| }
|
|
|
| -bool WebRtcVoiceEngine::SetLocalMonitor(bool enable) {
|
| - desired_local_monitor_enable_ = enable;
|
| - return ChangeLocalMonitor(desired_local_monitor_enable_);
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::ChangeLocalMonitor(bool enable) {
|
| - // The voe file api is not available in chrome.
|
| - if (!voe_wrapper_->file()) {
|
| - return false;
|
| - }
|
| - if (enable && !monitor_) {
|
| - monitor_.reset(new WebRtcMonitorStream);
|
| - if (voe_wrapper_->file()->StartRecordingMicrophone(monitor_.get()) == -1) {
|
| - LOG_RTCERR1(StartRecordingMicrophone, monitor_.get());
|
| - // Must call Stop() because there are some cases where Start will report
|
| - // failure but still change the state, and if we leave VE in the on state
|
| - // then it could crash later when trying to invoke methods on our monitor.
|
| - voe_wrapper_->file()->StopRecordingMicrophone();
|
| - monitor_.reset();
|
| - return false;
|
| - }
|
| - } else if (!enable && monitor_) {
|
| - voe_wrapper_->file()->StopRecordingMicrophone();
|
| - monitor_.reset();
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::PauseLocalMonitor() {
|
| - return ChangeLocalMonitor(false);
|
| -}
|
| -
|
| -bool WebRtcVoiceEngine::ResumeLocalMonitor() {
|
| - return ChangeLocalMonitor(desired_local_monitor_enable_);
|
| -}
|
| -
|
| const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
|
| return codecs_;
|
| }
|
|
|