| Index: webrtc/audio_receive_stream.h
|
| diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
|
| index 97feccc9386441752c61328b59349e332dda919f..6d72b4d3185037e6679faced29c8f0dc326b9627 100644
|
| --- a/webrtc/audio_receive_stream.h
|
| +++ b/webrtc/audio_receive_stream.h
|
| @@ -16,8 +16,8 @@
|
| #include <string>
|
| #include <vector>
|
|
|
| +#include "webrtc/common_types.h"
|
| #include "webrtc/config.h"
|
| -#include "webrtc/stream.h"
|
| #include "webrtc/transport.h"
|
| #include "webrtc/typedefs.h"
|
|
|
| @@ -31,7 +31,7 @@ class AudioSinkInterface;
|
| // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
|
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
|
|
|
| -class AudioReceiveStream : public ReceiveStream {
|
| +class AudioReceiveStream {
|
| public:
|
| struct Stats {
|
| uint32_t remote_ssrc = 0;
|
| @@ -103,6 +103,13 @@ class AudioReceiveStream : public ReceiveStream {
|
| std::map<uint8_t, AudioDecoder*> decoder_map;
|
| };
|
|
|
| + // Starts stream activity.
|
| + // When a stream is active, it can receive, process and deliver packets.
|
| + virtual void Start() = 0;
|
| + // Stops stream activity.
|
| + // When a stream is stopped, it can't receive, process or deliver packets.
|
| + virtual void Stop() = 0;
|
| +
|
| virtual Stats GetStats() const = 0;
|
|
|
| // Sets an audio sink that receives unmixed audio from the receive stream.
|
| @@ -114,6 +121,9 @@ class AudioReceiveStream : public ReceiveStream {
|
| // is being pulled+rendered and/or if audio is being pulled for the purposes
|
| // of feeding to the AEC.
|
| virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
|
| +
|
| + protected:
|
| + virtual ~AudioReceiveStream() {}
|
| };
|
| } // namespace webrtc
|
|
|
|
|