| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index 70f7f3a4b4c5a79e93a3dde73ab1e9858701c054..d2db76a7bdba63347740116c18984f428e073d28 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -1336,9 +1336,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
|
| return config_.voe_channel_id;
|
| }
|
|
|
| - void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
|
| + void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| - stream_->SetSink(rtc::ScopedToUnique(std::move(sink)));
|
| + stream_->SetSink(std::move(sink));
|
| }
|
|
|
| private:
|
| @@ -2262,7 +2262,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
|
| default_recv_ssrc_ = ssrc;
|
| SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
|
| if (default_sink_) {
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
| + std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
| new ProxySink(default_sink_.get()));
|
| SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
|
| }
|
| @@ -2492,13 +2492,13 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
|
|
| void WebRtcVoiceMediaChannel::SetRawAudioSink(
|
| uint32_t ssrc,
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
|
| + std::unique_ptr<webrtc::AudioSinkInterface> sink) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
|
| << " " << (sink ? "(ptr)" : "NULL");
|
| if (ssrc == 0) {
|
| if (default_recv_ssrc_ != -1) {
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
| + std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
| sink ? new ProxySink(sink.get()) : nullptr);
|
| SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
|
| }
|
|
|