Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(433)

Unified Diff: webrtc/audio/audio_receive_stream.h

Issue 1924793002: Remove webrtc/stream.h and unutilized inheritance. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: re-rebase Created 4 years, 8 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/DEPS ('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 37117427a172d11ad48a95cd414b2d985d4f79fb..d99956c2ddd381b447ce5b7c11679ad7a50d9fce 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -36,20 +36,17 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
~AudioReceiveStream() override;
- // webrtc::ReceiveStream implementation.
+ // webrtc::AudioReceiveStream implementation.
void Start() override;
void Stop() override;
- void SignalNetworkState(NetworkState state) override;
- bool DeliverRtcp(const uint8_t* packet, size_t length) override;
- bool DeliverRtp(const uint8_t* packet,
- size_t length,
- const PacketTime& packet_time) override;
-
- // webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
-
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
+ void SignalNetworkState(NetworkState state);
+ bool DeliverRtcp(const uint8_t* packet, size_t length);
+ bool DeliverRtp(const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time);
const webrtc::AudioReceiveStream::Config& config() const;
private:
« no previous file with comments | « webrtc/DEPS ('k') | webrtc/audio/audio_receive_stream.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698