273a414b0e
Implements reporting transmitted frame size in WebRtcVideoEngine2. R=mflodman@webrtc.org, stefan@webrtc.org BUG=4033 Review URL: https://webrtc-codereview.appspot.com/33399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
163 lines
5.1 KiB
C++
163 lines
5.1 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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media_bitrate_bps(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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int media_bitrate_bps;
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bool suspended;
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std::map<uint32_t, SsrcStats> 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() : max_packet_size(kDefaultMaxPacketSize) {}
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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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// 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 VideoEncoderConfig& config) = 0;
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virtual Stats GetStats() = 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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