Index: webrtc/call/video_receive_stream.h |
diff --git a/webrtc/video_receive_stream.h b/webrtc/call/video_receive_stream.h |
similarity index 94% |
copy from webrtc/video_receive_stream.h |
copy to webrtc/call/video_receive_stream.h |
index 48d16a8ea1b7f883d8652deceb694512c0946cfa..b1536f26e3fefc2d11ff90422abbd34bee65ea21 100644 |
--- a/webrtc/video_receive_stream.h |
+++ b/webrtc/call/video_receive_stream.h |
@@ -8,8 +8,8 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
-#ifndef WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
-#define WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
+#ifndef WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_ |
+#define WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_ |
#include <limits> |
#include <map> |
@@ -33,6 +33,9 @@ class VideoReceiveStream { |
// TODO(mflodman) Move all these settings to VideoDecoder and move the |
// declaration to common_types.h. |
struct Decoder { |
+ Decoder(); |
+ Decoder(const Decoder&); |
+ ~Decoder(); |
std::string ToString() const; |
// The actual decoder instance. |
@@ -53,6 +56,8 @@ class VideoReceiveStream { |
}; |
struct Stats { |
+ Stats(); |
+ ~Stats(); |
std::string ToString(int64_t time_ms) const; |
int network_frame_rate = 0; |
@@ -95,16 +100,15 @@ class VideoReceiveStream { |
private: |
// Access to the copy constructor is private to force use of the Copy() |
// method for those exceptional cases where we do use it. |
- Config(const Config&) = default; |
+ Config(const Config&); |
public: |
Config() = delete; |
- Config(Config&&) = default; |
- explicit Config(Transport* rtcp_send_transport) |
- : rtcp_send_transport(rtcp_send_transport) {} |
- |
- Config& operator=(Config&&) = default; |
+ Config(Config&&); |
+ explicit Config(Transport* rtcp_send_transport); |
+ Config& operator=(Config&&); |
Config& operator=(const Config&) = delete; |
+ ~Config(); |
// Mostly used by tests. Avoid creating copies if you can. |
Config Copy() const { return Config(*this); } |
@@ -116,6 +120,9 @@ class VideoReceiveStream { |
// Receive-stream specific RTP settings. |
struct Rtp { |
+ Rtp(); |
+ Rtp(const Rtp&); |
+ ~Rtp(); |
std::string ToString() const; |
// Synchronization source (stream identifier) to be received. |
@@ -235,4 +242,4 @@ class VideoReceiveStream { |
} // namespace webrtc |
-#endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
+#endif // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_ |