| 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_
|
|
|