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mflodman@webrtc.org06e80262013-04-18 12:02:521/*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
mflodman@webrtc.org5e0cbcf2013-12-18 09:46:2211#ifndef WEBRTC_VIDEO_SEND_STREAM_H_
12#define WEBRTC_VIDEO_SEND_STREAM_H_
mflodman@webrtc.org06e80262013-04-18 12:02:5213
sprang@webrtc.org49812e62014-01-07 09:54:3414#include <map>
mflodman@webrtc.org06e80262013-04-18 12:02:5215#include <string>
mflodman@webrtc.org06e80262013-04-18 12:02:5216
17#include "webrtc/common_types.h"
pbos@webrtc.org24e20892013-10-28 16:32:0118#include "webrtc/config.h"
19#include "webrtc/frame_callback.h"
20#include "webrtc/video_renderer.h"
mflodman@webrtc.org06e80262013-04-18 12:02:5221
22namespace webrtc {
23
24class VideoEncoder;
25
mflodman@webrtc.org06e80262013-04-18 12:02:5226// Class to deliver captured frame to the video send stream.
27class VideoSendStreamInput {
28 public:
pbos@webrtc.orgc33d37c2013-12-11 16:26:1629 // These methods do not lock internally and must be called sequentially.
30 // If your application switches input sources synchronization must be done
31 // externally to make sure that any old frames are not delivered concurrently.
pbos@webrtc.orgc33d37c2013-12-11 16:26:1632 virtual void SwapFrame(I420VideoFrame* video_frame) = 0;
mflodman@webrtc.org06e80262013-04-18 12:02:5233
34 protected:
35 virtual ~VideoSendStreamInput() {}
36};
37
mflodman@webrtc.org06e80262013-04-18 12:02:5238class VideoSendStream {
39 public:
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2140 struct Stats {
41 Stats()
42 : input_frame_rate(0),
sprang@webrtc.org49812e62014-01-07 09:54:3443 encode_frame_rate(0),
stefan@webrtc.org52322672014-11-05 14:05:2944 media_bitrate_bps(0),
henrik.lundin@webrtc.org9376c692014-03-13 13:31:2145 suspended(false) {}
sprang@webrtc.org49812e62014-01-07 09:54:3446 int input_frame_rate;
47 int encode_frame_rate;
stefan@webrtc.org52322672014-11-05 14:05:2948 int media_bitrate_bps;
henrik.lundin@webrtc.org9376c692014-03-13 13:31:2149 bool suspended;
stefan@webrtc.org52322672014-11-05 14:05:2950 std::map<uint32_t, SsrcStats> substreams;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2151 };
52
53 struct Config {
54 Config()
55 : pre_encode_callback(NULL),
sprang@webrtc.org2e98d452013-11-26 11:41:5956 post_encode_callback(NULL),
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2157 local_renderer(NULL),
58 render_delay_ms(0),
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2159 target_delay_ms(0),
henrik.lundin@webrtc.org45901772013-11-18 12:18:4360 suspend_below_min_bitrate(false) {}
pbos@webrtc.org7e686932014-05-15 09:35:0661 std::string ToString() const;
62
pbos@webrtc.orge2a7a772014-03-19 08:43:5763 struct EncoderSettings {
pbos@webrtc.orgbdfcddf2014-06-06 10:49:1964 EncoderSettings() : payload_type(-1), encoder(NULL) {}
pbos@webrtc.org23668752014-10-24 09:23:2165
pbos@webrtc.org7e686932014-05-15 09:35:0666 std::string ToString() const;
67
pbos@webrtc.orge2a7a772014-03-19 08:43:5768 std::string payload_name;
69 int payload_type;
70
71 // Uninitialized VideoEncoder instance to be used for encoding. Will be
72 // initialized from inside the VideoSendStream.
pbos@webrtc.orgf076e172015-01-15 10:09:3973 VideoEncoder* encoder;
pbos@webrtc.orge2a7a772014-03-19 08:43:5774 } encoder_settings;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2175
sprang@webrtc.org44bb62a2013-10-16 13:29:1476 static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2177 struct Rtp {
pbos@webrtc.org23668752014-10-24 09:23:2178 Rtp() : max_packet_size(kDefaultMaxPacketSize) {}
pbos@webrtc.org7e686932014-05-15 09:35:0679 std::string ToString() const;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:2180
81 std::vector<uint32_t> ssrcs;
82
83 // Max RTP packet size delivered to send transport from VideoEngine.
84 size_t max_packet_size;
85
86 // RTP header extensions to use for this send stream.
87 std::vector<RtpExtension> extensions;
88
89 // See NackConfig for description.
90 NackConfig nack;
91
92 // See FecConfig for description.
93 FecConfig fec;
94
pbos@webrtc.orgc71929d2014-01-24 09:30:5395 // Settings for RTP retransmission payload format, see RFC 4588 for
96 // details.
97 struct Rtx {
stefan@webrtc.org7c8ebe82014-12-19 15:33:1798 Rtx() : payload_type(-1) {}
pbos@webrtc.org7e686932014-05-15 09:35:0699 std::string ToString() const;
pbos@webrtc.orgc71929d2014-01-24 09:30:53100 // SSRCs to use for the RTX streams.
101 std::vector<uint32_t> ssrcs;
102
103 // Payload type to use for the RTX stream.
104 int payload_type;
105 } rtx;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21106
107 // RTCP CNAME, see RFC 3550.
108 std::string c_name;
109 } rtp;
110
111 // Called for each I420 frame before encoding the frame. Can be used for
112 // effects, snapshots etc. 'NULL' disables the callback.
113 I420FrameCallback* pre_encode_callback;
114
115 // Called for each encoded frame, e.g. used for file storage. 'NULL'
116 // disables the callback.
sprang@webrtc.org2e98d452013-11-26 11:41:59117 EncodedFrameObserver* post_encode_callback;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21118
119 // Renderer for local preview. The local renderer will be called even if
120 // sending hasn't started. 'NULL' disables local rendering.
121 VideoRenderer* local_renderer;
122
123 // Expected delay needed by the renderer, i.e. the frame will be delivered
124 // this many milliseconds, if possible, earlier than expected render time.
pbos@webrtc.org7e686932014-05-15 09:35:06125 // Only valid if |local_renderer| is set.
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21126 int render_delay_ms;
127
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21128 // Target delay in milliseconds. A positive value indicates this stream is
129 // used for streaming instead of a real-time call.
130 int target_delay_ms;
131
henrik.lundin@webrtc.org45901772013-11-18 12:18:43132 // True if the stream should be suspended when the available bitrate fall
133 // below the minimum configured bitrate. If this variable is false, the
134 // stream may send at a rate higher than the estimated available bitrate.
135 bool suspend_below_min_bitrate;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21136 };
137
mflodman@webrtc.org06e80262013-04-18 12:02:52138 // Gets interface used to insert captured frames. Valid as long as the
139 // VideoSendStream is valid.
140 virtual VideoSendStreamInput* Input() = 0;
141
pbos@webrtc.org16a058a2014-04-24 11:13:21142 virtual void Start() = 0;
143 virtual void Stop() = 0;
mflodman@webrtc.org06e80262013-04-18 12:02:52144
pbos@webrtc.orge2a7a772014-03-19 08:43:57145 // Set which streams to send. Must have at least as many SSRCs as configured
146 // in the config. Encoder settings are passed on to the encoder instance along
147 // with the VideoStream settings.
pbos@webrtc.org58b51402014-09-19 12:30:25148 virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
mflodman@webrtc.org06e80262013-04-18 12:02:52149
pbos@webrtc.orgee9497c2014-12-01 15:23:21150 virtual Stats GetStats() = 0;
sprang@webrtc.org49812e62014-01-07 09:54:34151
mflodman@webrtc.org06e80262013-04-18 12:02:52152 protected:
153 virtual ~VideoSendStream() {}
154};
155
mflodman@webrtc.org06e80262013-04-18 12:02:52156} // namespace webrtc
157
mflodman@webrtc.org5e0cbcf2013-12-18 09:46:22158#endif // WEBRTC_VIDEO_SEND_STREAM_H_