This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems. R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21869005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
		
			
				
	
	
		
			168 lines
		
	
	
		
			5.4 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			168 lines
		
	
	
		
			5.4 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
/*
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 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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 *
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 *  Use of this source code is governed by a BSD-style license
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 *  that can be found in the LICENSE file in the root of the source
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 *  tree. An additional intellectual property rights grant can be found
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 *  in the file PATENTS.  All contributing project authors may
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 *  be found in the AUTHORS file in the root of the source tree.
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 */
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#ifndef WEBRTC_VIDEO_SEND_STREAM_H_
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#define WEBRTC_VIDEO_SEND_STREAM_H_
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#include <map>
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#include <string>
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#include "webrtc/common_types.h"
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#include "webrtc/config.h"
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#include "webrtc/frame_callback.h"
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#include "webrtc/video_renderer.h"
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namespace webrtc {
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class VideoEncoder;
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// Class to deliver captured frame to the video send stream.
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class VideoSendStreamInput {
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 public:
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  // These methods do not lock internally and must be called sequentially.
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  // If your application switches input sources synchronization must be done
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  // externally to make sure that any old frames are not delivered concurrently.
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  virtual void SwapFrame(I420VideoFrame* video_frame) = 0;
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 protected:
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  virtual ~VideoSendStreamInput() {}
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};
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class VideoSendStream {
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 public:
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  struct Stats {
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    Stats()
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        : input_frame_rate(0),
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          encode_frame_rate(0),
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          suspended(false) {}
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    int input_frame_rate;
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    int encode_frame_rate;
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    bool suspended;
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    std::map<uint32_t, StreamStats> substreams;
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  };
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  struct Config {
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    Config()
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        : pre_encode_callback(NULL),
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          post_encode_callback(NULL),
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          local_renderer(NULL),
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          render_delay_ms(0),
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          target_delay_ms(0),
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          suspend_below_min_bitrate(false) {}
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    std::string ToString() const;
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    struct EncoderSettings {
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      EncoderSettings() : payload_type(-1), encoder(NULL) {}
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      std::string ToString() const;
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      std::string payload_name;
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      int payload_type;
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      // Uninitialized VideoEncoder instance to be used for encoding. Will be
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      // initialized from inside the VideoSendStream.
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      webrtc::VideoEncoder* encoder;
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    } encoder_settings;
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    static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
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    struct Rtp {
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      Rtp()
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          : max_packet_size(kDefaultMaxPacketSize),
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            min_transmit_bitrate_bps(0) {}
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      std::string ToString() const;
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      std::vector<uint32_t> ssrcs;
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      // Max RTP packet size delivered to send transport from VideoEngine.
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      size_t max_packet_size;
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      // Padding will be used up to this bitrate regardless of the bitrate
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      // produced by the encoder. Padding above what's actually produced by the
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      // encoder helps maintaining a higher bitrate estimate.
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      int min_transmit_bitrate_bps;
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      // RTP header extensions to use for this send stream.
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      std::vector<RtpExtension> extensions;
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      // See NackConfig for description.
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      NackConfig nack;
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      // See FecConfig for description.
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      FecConfig fec;
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      // Settings for RTP retransmission payload format, see RFC 4588 for
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      // details.
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      struct Rtx {
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        Rtx() : payload_type(-1), pad_with_redundant_payloads(false) {}
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        std::string ToString() const;
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        // SSRCs to use for the RTX streams.
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        std::vector<uint32_t> ssrcs;
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        // Payload type to use for the RTX stream.
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        int payload_type;
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        // Use redundant payloads to pad the bitrate. Instead of padding with
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        // randomized packets, we will preemptively retransmit media packets on
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        // the RTX stream.
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        bool pad_with_redundant_payloads;
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      } rtx;
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      // RTCP CNAME, see RFC 3550.
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      std::string c_name;
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    } rtp;
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    // Called for each I420 frame before encoding the frame. Can be used for
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    // effects, snapshots etc. 'NULL' disables the callback.
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    I420FrameCallback* pre_encode_callback;
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    // Called for each encoded frame, e.g. used for file storage. 'NULL'
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    // disables the callback.
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    EncodedFrameObserver* post_encode_callback;
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    // Renderer for local preview. The local renderer will be called even if
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    // sending hasn't started. 'NULL' disables local rendering.
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    VideoRenderer* local_renderer;
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    // Expected delay needed by the renderer, i.e. the frame will be delivered
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    // this many milliseconds, if possible, earlier than expected render time.
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    // Only valid if |local_renderer| is set.
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    int render_delay_ms;
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    // Target delay in milliseconds. A positive value indicates this stream is
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    // used for streaming instead of a real-time call.
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    int target_delay_ms;
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    // True if the stream should be suspended when the available bitrate fall
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    // below the minimum configured bitrate. If this variable is false, the
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    // stream may send at a rate higher than the estimated available bitrate.
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    bool suspend_below_min_bitrate;
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  };
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  // Gets interface used to insert captured frames. Valid as long as the
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  // VideoSendStream is valid.
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  virtual VideoSendStreamInput* Input() = 0;
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  virtual void Start() = 0;
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  virtual void Stop() = 0;
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  // Set which streams to send. Must have at least as many SSRCs as configured
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  // in the config. Encoder settings are passed on to the encoder instance along
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  // with the VideoStream settings.
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  virtual bool ReconfigureVideoEncoder(const std::vector<VideoStream>& streams,
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                                       const void* encoder_settings) = 0;
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  virtual Stats GetStats() const = 0;
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 protected:
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  virtual ~VideoSendStream() {}
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};
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}  // namespace webrtc
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#endif  // WEBRTC_VIDEO_SEND_STREAM_H_
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