Move video send/receive stream headers to webrtc/call.

webrtc/video_receive_stream.h

Index: webrtc/video_receive_stream.h
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index 48d16a8ea1b7f883d8652deceb694512c0946cfa..86d71d3f11264dea70035b77569ecd648d06b753 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -11,228 +11,8 @@
 #ifndef WEBRTC_VIDEO_RECEIVE_STREAM_H_
 #define WEBRTC_VIDEO_RECEIVE_STREAM_H_
 
-#include <limits>
-#include <map>
-#include <string>
-#include <vector>
-
-#include "webrtc/api/call/transport.h"
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/config.h"
-#include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/rtc_base/platform_file.h"
-
-namespace webrtc {
-
-class RtpPacketSinkInterface;
-class VideoDecoder;
-
-class VideoReceiveStream {
- public:
-  // TODO(mflodman) Move all these settings to VideoDecoder and move the
-  // declaration to common_types.h.
-  struct Decoder {
-    std::string ToString() const;
-
-    // The actual decoder instance.
-    VideoDecoder* decoder = nullptr;
-
-    // Received RTP packets with this payload type will be sent to this decoder
-    // instance.
-    int payload_type = 0;
-
-    // Name of the decoded payload (such as VP8). Maps back to the depacketizer
-    // used to unpack incoming packets.
-    std::string payload_name;
-
-    // This map contains the codec specific parameters from SDP, i.e. the "fmtp"
-    // parameters. It is the same as cricket::CodecParameterMap used in
-    // cricket::VideoCodec.
-    std::map<std::string, std::string> codec_params;
-  };
-
-  struct Stats {
-    std::string ToString(int64_t time_ms) const;
-
-    int network_frame_rate = 0;
-    int decode_frame_rate = 0;
-    int render_frame_rate = 0;
-    uint32_t frames_rendered = 0;
-
-    // Decoder stats.
-    std::string decoder_implementation_name = "unknown";
-    FrameCounts frame_counts;
-    int decode_ms = 0;
-    int max_decode_ms = 0;
-    int current_delay_ms = 0;
-    int target_delay_ms = 0;
-    int jitter_buffer_ms = 0;
-    int min_playout_delay_ms = 0;
-    int render_delay_ms = 10;
-    uint64_t interframe_delay_sum_ms = 0;
-    uint32_t frames_decoded = 0;
-    rtc::Optional<uint64_t> qp_sum;
-
-    int current_payload_type = -1;
-
-    int total_bitrate_bps = 0;
-    int discarded_packets = 0;
-
-    int width = 0;
-    int height = 0;
-
-    int sync_offset_ms = std::numeric_limits<int>::max();
-
-    uint32_t ssrc = 0;
-    std::string c_name;
-    StreamDataCounters rtp_stats;
-    RtcpPacketTypeCounter rtcp_packet_type_counts;
-    RtcpStatistics rtcp_stats;
-  };
-
-  struct Config {
-   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;
-
-   public:
-    Config() = delete;
-    Config(Config&&) = default;
-    explicit Config(Transport* rtcp_send_transport)
-        : rtcp_send_transport(rtcp_send_transport) {}
-
-    Config& operator=(Config&&) = default;
-    Config& operator=(const Config&) = delete;
-
-    // Mostly used by tests.  Avoid creating copies if you can.
-    Config Copy() const { return Config(*this); }
-
-    std::string ToString() const;
-
-    // Decoders for every payload that we can receive.
-    std::vector<Decoder> decoders;
-
-    // Receive-stream specific RTP settings.
-    struct Rtp {
-      std::string ToString() const;
-
-      // Synchronization source (stream identifier) to be received.
-      uint32_t remote_ssrc = 0;
-
-      // Sender SSRC used for sending RTCP (such as receiver reports).
-      uint32_t local_ssrc = 0;
-
-      // See RtcpMode for description.
-      RtcpMode rtcp_mode = RtcpMode::kCompound;
-
-      // Extended RTCP settings.
-      struct RtcpXr {
-        // True if RTCP Receiver Reference Time Report Block extension
-        // (RFC 3611) should be enabled.
-        bool receiver_reference_time_report = false;
-      } rtcp_xr;
-
-      // TODO(nisse): This remb setting is currently set but never
-      // applied. REMB logic is now the responsibility of
-      // PacketRouter, and it will generate REMB feedback if
-      // OnReceiveBitrateChanged is used, which depends on how the
-      // estimators belonging to the ReceiveSideCongestionController
-      // are configured. Decide if this setting should be deleted, and
-      // if it needs to be replaced by a setting in PacketRouter to
-      // disable REMB feedback.
-
-      // See draft-alvestrand-rmcat-remb for information.
-      bool remb = false;
-
-      // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
-      bool transport_cc = false;
-
-      // See NackConfig for description.
-      NackConfig nack;
-
-      // See UlpfecConfig for description.
-      UlpfecConfig ulpfec;
-
-      // SSRC for retransmissions.
-      uint32_t rtx_ssrc = 0;
-
-      // Set if the stream is protected using FlexFEC.
-      bool protected_by_flexfec = false;
-
-      // Map from video payload type (apt) -> RTX payload type (pt).
-      // For RTX to be enabled, both an SSRC and this mapping are needed.
-      std::map<int, int> rtx_payload_types;
-
-      // RTP header extensions used for the received stream.
-      std::vector<RtpExtension> extensions;
-    } rtp;
-
-    // Transport for outgoing packets (RTCP).
-    Transport* rtcp_send_transport = nullptr;
-
-    // Must not be 'nullptr' when the stream is started.
-    rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
-
-    // Expected delay needed by the renderer, i.e. the frame will be delivered
-    // this many milliseconds, if possible, earlier than the ideal render time.
-    // Only valid if 'renderer' is set.
-    int render_delay_ms = 10;
-
-    // If set, pass frames on to the renderer as soon as they are
-    // available.
-    bool disable_prerenderer_smoothing = false;
-
-    // Identifier for an A/V synchronization group. Empty string to disable.
-    // TODO(pbos): Synchronize streams in a sync group, not just video streams
-    // to one of the audio streams.
-    std::string sync_group;
-
-    // Called for each incoming video frame, i.e. in encoded state. E.g. used
-    // when
-    // saving the stream to a file. 'nullptr' disables the callback.
-    EncodedFrameObserver* pre_decode_callback = nullptr;
-
-    // Target delay in milliseconds. A positive value indicates this stream is
-    // used for streaming instead of a real-time call.
-    int target_delay_ms = 0;
-  };
-
-  // 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;
-
-  // TODO(pbos): Add info on currently-received codec to Stats.
-  virtual Stats GetStats() const = 0;
-
-  virtual rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() = 0;
-
-  // Takes ownership of the file, is responsible for closing it later.
-  // Calling this method will close and finalize any current log.
-  // Giving rtc::kInvalidPlatformFileValue disables logging.
-  // If a frame to be written would make the log too large the write fails and
-  // the log is closed and finalized. A |byte_limit| of 0 means no limit.
-  virtual void EnableEncodedFrameRecording(rtc::PlatformFile file,
-                                           size_t byte_limit) = 0;
-  inline void DisableEncodedFrameRecording() {
-    EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0);
-  }
-
-  // RtpDemuxer only forwards a given RTP packet to one sink. However, some
-  // sinks, such as FlexFEC, might wish to be informed of all of the packets
-  // a given sink receives (or any set of sinks). They may do so by registering
-  // themselves as secondary sinks.
-  virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
-  virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
-
- protected:
-  virtual ~VideoReceiveStream() {}
-};
-
-}  // namespace webrtc
+#include "webrtc/call/video_receive_stream.h"
+// The contents header have moved to webrtc/call/video_receive_stream.h. This
+// file is deprecated. See http://bugs.webrtc.org/8107.
 
 #endif  // WEBRTC_VIDEO_RECEIVE_STREAM_H_