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

webrtc/video_send_stream.h

Index: webrtc/video_send_stream.h
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index c5a1f9b647ace77a4004eccaa161ca6ee28eced0..74be574f87318a03f958745b441d72b07eab7250 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -11,258 +11,8 @@
 #ifndef WEBRTC_VIDEO_SEND_STREAM_H_
 #define WEBRTC_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"
-#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/config.h"
-#include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/media/base/videosourceinterface.h"
-#include "webrtc/rtc_base/platform_file.h"
-
-namespace webrtc {
-
-class VideoEncoder;
-
-class VideoSendStream {
- public:
-  struct StreamStats {
-    std::string ToString() const;
-
-    FrameCounts frame_counts;
-    bool is_rtx = false;
-    bool is_flexfec = false;
-    int width = 0;
-    int height = 0;
-    // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
-    int total_bitrate_bps = 0;
-    int retransmit_bitrate_bps = 0;
-    int avg_delay_ms = 0;
-    int max_delay_ms = 0;
-    StreamDataCounters rtp_stats;
-    RtcpPacketTypeCounter rtcp_packet_type_counts;
-    RtcpStatistics rtcp_stats;
-  };
-
-  struct Stats {
-    std::string ToString(int64_t time_ms) const;
-    std::string encoder_implementation_name = "unknown";
-    int input_frame_rate = 0;
-    int encode_frame_rate = 0;
-    int avg_encode_time_ms = 0;
-    int encode_usage_percent = 0;
-    uint32_t frames_encoded = 0;
-    rtc::Optional<uint64_t> qp_sum;
-    // Bitrate the encoder is currently configured to use due to bandwidth
-    // limitations.
-    int target_media_bitrate_bps = 0;
-    // Bitrate the encoder is actually producing.
-    int media_bitrate_bps = 0;
-    // Media bitrate this VideoSendStream is configured to prefer if there are
-    // no bandwidth limitations.
-    int preferred_media_bitrate_bps = 0;
-    bool suspended = false;
-    bool bw_limited_resolution = false;
-    bool cpu_limited_resolution = false;
-    bool bw_limited_framerate = false;
-    bool cpu_limited_framerate = false;
-    // Total number of times resolution as been requested to be changed due to
-    // CPU/quality adaptation.
-    int number_of_cpu_adapt_changes = 0;
-    int number_of_quality_adapt_changes = 0;
-    std::map<uint32_t, StreamStats> substreams;
-  };
-
-  struct Config {
-   public:
-    Config() = delete;
-    Config(Config&&) = default;
-    explicit Config(Transport* send_transport)
-        : send_transport(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;
-
-    struct EncoderSettings {
-      EncoderSettings() = default;
-      EncoderSettings(std::string payload_name,
-                      int payload_type,
-                      VideoEncoder* encoder)
-          : payload_name(std::move(payload_name)),
-            payload_type(payload_type),
-            encoder(encoder) {}
-      std::string ToString() const;
-
-      std::string payload_name;
-      int payload_type = -1;
-
-      // TODO(sophiechang): Delete this field when no one is using internal
-      // sources anymore.
-      bool internal_source = false;
-
-      // Allow 100% encoder utilization. Used for HW encoders where CPU isn't
-      // expected to be the limiting factor, but a chip could be running at
-      // 30fps (for example) exactly.
-      bool full_overuse_time = false;
-
-      // Uninitialized VideoEncoder instance to be used for encoding. Will be
-      // initialized from inside the VideoSendStream.
-      VideoEncoder* encoder = nullptr;
-    } encoder_settings;
-
-    static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
-    struct Rtp {
-      std::string ToString() const;
-
-      std::vector<uint32_t> ssrcs;
-
-      // See RtcpMode for description.
-      RtcpMode rtcp_mode = RtcpMode::kCompound;
-
-      // Max RTP packet size delivered to send transport from VideoEngine.
-      size_t max_packet_size = kDefaultMaxPacketSize;
-
-      // RTP header extensions to use for this send stream.
-      std::vector<RtpExtension> extensions;
-
-      // See NackConfig for description.
-      NackConfig nack;
-
-      // See UlpfecConfig for description.
-      UlpfecConfig ulpfec;
-
-      struct Flexfec {
-        // Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
-        int payload_type = -1;
-
-        // SSRC of FlexFEC stream.
-        uint32_t ssrc = 0;
-
-        // Vector containing a single element, corresponding to the SSRC of the
-        // media stream being protected by this FlexFEC stream.
-        // The vector MUST have size 1.
-        //
-        // TODO(brandtr): Update comment above when we support
-        // multistream protection.
-        std::vector<uint32_t> protected_media_ssrcs;
-      } flexfec;
-
-      // Settings for RTP retransmission payload format, see RFC 4588 for
-      // details.
-      struct Rtx {
-        std::string ToString() const;
-        // SSRCs to use for the RTX streams.
-        std::vector<uint32_t> ssrcs;
-
-        // Payload type to use for the RTX stream.
-        int payload_type = -1;
-      } rtx;
-
-      // RTCP CNAME, see RFC 3550.
-      std::string c_name;
-    } rtp;
-
-    // Transport for outgoing packets.
-    Transport* send_transport = nullptr;
-
-    // Called for each I420 frame before encoding the frame. Can be used for
-    // effects, snapshots etc. 'nullptr' disables the callback.
-    rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
-
-    // Called for each encoded frame, e.g. used for file storage. 'nullptr'
-    // disables the callback. Also measures timing and passes the time
-    // spent on encoding. This timing will not fire if encoding takes longer
-    // than the measuring window, since the sample data will have been dropped.
-    EncodedFrameObserver* post_encode_callback = nullptr;
-
-    // Expected delay needed by the renderer, i.e. the frame will be delivered
-    // this many milliseconds, if possible, earlier than expected render time.
-    // Only valid if |local_renderer| is set.
-    int render_delay_ms = 0;
-
-    // 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;
-
-    // True if the stream should be suspended when the available bitrate fall
-    // below the minimum configured bitrate. If this variable is false, the
-    // stream may send at a rate higher than the estimated available bitrate.
-    bool suspend_below_min_bitrate = false;
-
-    // Enables periodic bandwidth probing in application-limited region.
-    bool periodic_alr_bandwidth_probing = false;
-
-   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;
-  };
-
-  // 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;
-
-  // Based on the spec in
-  // https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
-  // These options are enforced on a best-effort basis. For instance, all of
-  // these options may suffer some frame drops in order to avoid queuing.
-  // TODO(sprang): Look into possibility of more strictly enforcing the
-  // maintain-framerate option.
-  enum class DegradationPreference {
-    // Don't take any actions based on over-utilization signals.
-    kDegradationDisabled,
-    // On over-use, request lower frame rate, possibly causing frame drops.
-    kMaintainResolution,
-    // On over-use, request lower resolution, possibly causing down-scaling.
-    kMaintainFramerate,
-    // Try to strike a "pleasing" balance between frame rate or resolution.
-    kBalanced,
-  };
-
-  virtual void SetSource(
-      rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
-      const DegradationPreference& degradation_preference) = 0;
-
-  // Set which streams to send. Must have at least as many SSRCs as configured
-  // in the config. Encoder settings are passed on to the encoder instance along
-  // with the VideoStream settings.
-  virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
-
-  virtual Stats GetStats() = 0;
-
-  // Takes ownership of each file, is responsible for closing them later.
-  // Calling this method will close and finalize any current logs.
-  // Some codecs produce multiple streams (VP8 only at present), each of these
-  // streams will log to a separate file. kMaxSimulcastStreams in common_types.h
-  // gives the max number of such streams. If there is no file for a stream, or
-  // the file is rtc::kInvalidPlatformFileValue, frames from that stream will
-  // not be logged.
-  // 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(
-      const std::vector<rtc::PlatformFile>& files,
-      size_t byte_limit) = 0;
-  inline void DisableEncodedFrameRecording() {
-    EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
-  }
-
- protected:
-  virtual ~VideoSendStream() {}
-};
-
-}  // namespace webrtc
+#include "webrtc/call/video_send_stream.h"
+// The contents header have moved to webrtc/call/video_send_stream.h. This
+// file is deprecated. See http://bugs.webrtc.org/8107.
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_