Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(188)

Unified Diff: webrtc/audio/audio_receive_stream.h

Issue 1390753002: Implement AudioReceiveStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: merge master Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/audio/BUILD.gn ('k') | webrtc/audio/audio_receive_stream.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/audio/audio_receive_stream.h
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 1e5272402026296e04c4aa2e807f9d9f3faaa027..5c77653a75425eb6e5f7560088b0e2b6eedb100a 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -12,18 +12,23 @@
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
#include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {
class RemoteBitrateEstimator;
+class VoiceEngine;
namespace internal {
class AudioReceiveStream : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
- const webrtc::AudioReceiveStream::Config& config);
+ const webrtc::AudioReceiveStream::Config& config,
+ VoiceEngine* voice_engine);
~AudioReceiveStream() override;
// webrtc::ReceiveStream implementation.
@@ -41,8 +46,12 @@ class AudioReceiveStream : public webrtc::AudioReceiveStream {
const webrtc::AudioReceiveStream::Config& config() const;
private:
+ rtc::ThreadChecker thread_checker_;
RemoteBitrateEstimator* const remote_bitrate_estimator_;
const webrtc::AudioReceiveStream::Config config_;
+ VoiceEngine* voice_engine_;
+ // We hold one interface pointer to the VoE to make sure it is kept alive.
+ ScopedVoEInterface<VoEBase> voe_base_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
};
} // namespace internal
« no previous file with comments | « webrtc/audio/BUILD.gn ('k') | webrtc/audio/audio_receive_stream.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698