webrtc/video_send_stream.h

/*
 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_SEND_STREAM_H_
#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_SEND_STREAM_H_

#include <string>
#include <vector>

#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/frame_callback.h"
#include "webrtc/video_renderer.h"

namespace webrtc {

class VideoEncoder;

struct SendStreamState;

// Class to deliver captured frame to the video send stream.
class VideoSendStreamInput {
 public:
  // TODO(mflodman) Replace time_since_capture_ms when I420VideoFrame uses NTP
  // time.
  virtual void PutFrame(const I420VideoFrame& video_frame,
                        uint32_t time_since_capture_ms) = 0;

 protected:
  virtual ~VideoSendStreamInput() {}
};

class VideoSendStream {
 public:
  struct Stats {
    Stats()
        : input_frame_rate(0),
          encode_frame(0),
          key_frames(0),
          delta_frames(0),
          video_packets(0),
          retransmitted_packets(0),
          fec_packets(0),
          padding_packets(0),
          send_bitrate_bps(0),
          delay_ms(0) {}
    RtpStatistics rtp;
    int input_frame_rate;
    int encode_frame;
    uint32_t key_frames;
    uint32_t delta_frames;
    uint32_t video_packets;
    uint32_t retransmitted_packets;
    uint32_t fec_packets;
    uint32_t padding_packets;
    int32_t send_bitrate_bps;
    int delay_ms;
  };

  class StatsCallback {
   public:
    virtual ~StatsCallback() {}
    virtual void ReceiveStats(const std::vector<Stats>& stats) = 0;
  };

  struct Config {
    Config()
        : pre_encode_callback(NULL),
          encoded_callback(NULL),
          local_renderer(NULL),
          render_delay_ms(0),
          encoder(NULL),
          internal_source(false),
          target_delay_ms(0),
          pacing(false),
          stats_callback(NULL),
          start_state(NULL),
          auto_mute(false) {}
    VideoCodec codec;

    static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
    struct Rtp {
      Rtp() : max_packet_size(kDefaultMaxPacketSize) {}

      std::vector<uint32_t> ssrcs;

      // Max RTP packet size delivered to send transport from VideoEngine.
      size_t max_packet_size;

      // RTP header extensions to use for this send stream.
      std::vector<RtpExtension> extensions;

      // See NackConfig for description.
      NackConfig nack;

      // See FecConfig for description.
      FecConfig fec;

      // See RtxConfig for description.
      RtxConfig rtx;

      // RTCP CNAME, see RFC 3550.
      std::string c_name;
    } rtp;

    // Called for each I420 frame before encoding the frame. Can be used for
    // effects, snapshots etc. 'NULL' disables the callback.
    I420FrameCallback* pre_encode_callback;

    // Called for each encoded frame, e.g. used for file storage. 'NULL'
    // disables the callback.
    EncodedFrameObserver* encoded_callback;

    // Renderer for local preview. The local renderer will be called even if
    // sending hasn't started. 'NULL' disables local rendering.
    VideoRenderer* local_renderer;

    // 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 |renderer| is set.
    int render_delay_ms;

    // TODO(mflodman) Move VideoEncoder to common_types.h and redefine.
    // External encoding. 'encoder' is the external encoder instance and
    // 'internal_source' is set to true if the encoder also captures the video
    // frames.
    VideoEncoder* encoder;
    bool internal_source;

    // Target delay in milliseconds. A positive value indicates this stream is
    // used for streaming instead of a real-time call.
    int target_delay_ms;

    // True if network a send-side packet buffer should be used to pace out
    // packets onto the network.
    bool pacing;

    // Callback for periodically receiving send stats.
    StatsCallback* stats_callback;

    // Set to resume a previously destroyed send stream.
    SendStreamState* start_state;

    // True if video should be muted when video goes under the minimum video
    // bitrate. Unless muted, video will be sent at a bitrate higher than
    // estimated available.
    bool auto_mute;
  };

  // Gets interface used to insert captured frames. Valid as long as the
  // VideoSendStream is valid.
  virtual VideoSendStreamInput* Input() = 0;

  virtual void StartSend() = 0;
  virtual void StopSend() = 0;

  // TODO(mflodman) Change VideoCodec struct and use here.
  virtual bool SetTargetBitrate(
      int min_bitrate, int max_bitrate,
      const std::vector<SimulcastStream>& streams) = 0;

  virtual void GetSendCodec(VideoCodec* send_codec) = 0;

 protected:
  virtual ~VideoSendStream() {}
};

}  // namespace webrtc

#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_SEND_STREAM_H_