Index: webrtc/call/video_send_stream.h |
diff --git a/webrtc/video_send_stream.h b/webrtc/call/video_send_stream.h |
similarity index 94% |
copy from webrtc/video_send_stream.h |
copy to webrtc/call/video_send_stream.h |
index c5a1f9b647ace77a4004eccaa161ca6ee28eced0..a176709ac46b6aaf92616d830c3020b5ca5e17d7 100644 |
--- a/webrtc/video_send_stream.h |
+++ b/webrtc/call/video_send_stream.h |
@@ -8,14 +8,13 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
-#ifndef WEBRTC_VIDEO_SEND_STREAM_H_ |
-#define WEBRTC_VIDEO_SEND_STREAM_H_ |
+#ifndef WEBRTC_CALL_VIDEO_SEND_STREAM_H_ |
+#define WEBRTC_CALL_VIDEO_SEND_STREAM_H_ |
#include <map> |
#include <string> |
#include <utility> |
#include <vector> |
-#include <utility> |
#include "webrtc/api/call/transport.h" |
#include "webrtc/common_types.h" |
@@ -32,6 +31,9 @@ class VideoEncoder; |
class VideoSendStream { |
public: |
struct StreamStats { |
+ StreamStats(); |
+ ~StreamStats(); |
+ |
std::string ToString() const; |
FrameCounts frame_counts; |
@@ -50,6 +52,8 @@ class VideoSendStream { |
}; |
struct Stats { |
+ Stats(); |
+ ~Stats(); |
std::string ToString(int64_t time_ms) const; |
std::string encoder_implementation_name = "unknown"; |
int input_frame_rate = 0; |
@@ -81,13 +85,14 @@ class VideoSendStream { |
struct Config { |
public: |
Config() = delete; |
- Config(Config&&) = default; |
- explicit Config(Transport* send_transport) |
- : send_transport(send_transport) {} |
+ Config(Config&&); |
+ explicit Config(Transport* send_transport); |
- Config& operator=(Config&&) = default; |
+ 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); } |
@@ -122,6 +127,9 @@ class VideoSendStream { |
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4. |
struct Rtp { |
+ Rtp(); |
+ Rtp(const Rtp&); |
+ ~Rtp(); |
std::string ToString() const; |
std::vector<uint32_t> ssrcs; |
@@ -142,6 +150,9 @@ class VideoSendStream { |
UlpfecConfig ulpfec; |
struct Flexfec { |
+ Flexfec(); |
+ Flexfec(const Flexfec&); |
+ ~Flexfec(); |
// Payload type of FlexFEC. Set to -1 to disable sending FlexFEC. |
int payload_type = -1; |
@@ -160,6 +171,9 @@ class VideoSendStream { |
// Settings for RTP retransmission payload format, see RFC 4588 for |
// details. |
struct Rtx { |
+ Rtx(); |
+ Rtx(const Rtx&); |
+ ~Rtx(); |
std::string ToString() const; |
// SSRCs to use for the RTX streams. |
std::vector<uint32_t> ssrcs; |
@@ -205,7 +219,7 @@ class VideoSendStream { |
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&); |
}; |
// Starts stream activity. |
@@ -265,4 +279,4 @@ class VideoSendStream { |
} // namespace webrtc |
-#endif // WEBRTC_VIDEO_SEND_STREAM_H_ |
+#endif // WEBRTC_CALL_VIDEO_SEND_STREAM_H_ |