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mflodman@webrtc.org65f995a2013-04-18 12:02:52 +00001/*
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.orgb429e512013-12-18 09:46:22 +000011#ifndef WEBRTC_VIDEO_SEND_STREAM_H_
12#define WEBRTC_VIDEO_SEND_STREAM_H_
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000013
sprang@webrtc.orgccd42842014-01-07 09:54:34 +000014#include <map>
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000015#include <string>
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000016
17#include "webrtc/common_types.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000018#include "webrtc/config.h"
19#include "webrtc/frame_callback.h"
20#include "webrtc/video_renderer.h"
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000021
22namespace webrtc {
23
24class VideoEncoder;
25
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000026// Class to deliver captured frame to the video send stream.
27class VideoSendStreamInput {
28 public:
pbos@webrtc.org724947b2013-12-11 16:26:16 +000029 // 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.org724947b2013-12-11 16:26:16 +000032 virtual void SwapFrame(I420VideoFrame* video_frame) = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000033
34 protected:
35 virtual ~VideoSendStreamInput() {}
36};
37
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000038class VideoSendStream {
39 public:
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000040 struct StreamStats {
41 FrameCounts frame_counts;
42 int width = 0;
43 int height = 0;
44 // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
45 int total_bitrate_bps = 0;
46 int retransmit_bitrate_bps = 0;
47 int avg_delay_ms = 0;
48 int max_delay_ms = 0;
49 StreamDataCounters rtp_stats;
50 RtcpPacketTypeCounter rtcp_packet_type_counts;
51 RtcpStatistics rtcp_stats;
52 };
53
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000054 struct Stats {
55 Stats()
56 : input_frame_rate(0),
sprang@webrtc.orgccd42842014-01-07 09:54:34 +000057 encode_frame_rate(0),
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +000058 media_bitrate_bps(0),
henrik.lundin@webrtc.orgb10363f2014-03-13 13:31:21 +000059 suspended(false) {}
sprang@webrtc.orgccd42842014-01-07 09:54:34 +000060 int input_frame_rate;
61 int encode_frame_rate;
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +000062 int media_bitrate_bps;
henrik.lundin@webrtc.orgb10363f2014-03-13 13:31:21 +000063 bool suspended;
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000064 std::map<uint32_t, StreamStats> substreams;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000065 };
66
67 struct Config {
68 Config()
69 : pre_encode_callback(NULL),
sprang@webrtc.org40709352013-11-26 11:41:59 +000070 post_encode_callback(NULL),
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000071 local_renderer(NULL),
72 render_delay_ms(0),
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000073 target_delay_ms(0),
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +000074 suspend_below_min_bitrate(false) {}
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000075 std::string ToString() const;
76
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000077 struct EncoderSettings {
pbos@webrtc.org6ae48c62014-06-06 10:49:19 +000078 EncoderSettings() : payload_type(-1), encoder(NULL) {}
pbos@webrtc.orgad3b5a52014-10-24 09:23:21 +000079
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000080 std::string ToString() const;
81
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000082 std::string payload_name;
83 int payload_type;
84
85 // Uninitialized VideoEncoder instance to be used for encoding. Will be
86 // initialized from inside the VideoSendStream.
pbos@webrtc.org32e85282015-01-15 10:09:39 +000087 VideoEncoder* encoder;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000088 } encoder_settings;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000089
sprang@webrtc.org25fce9a2013-10-16 13:29:14 +000090 static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000091 struct Rtp {
pbos@webrtc.orgad3b5a52014-10-24 09:23:21 +000092 Rtp() : max_packet_size(kDefaultMaxPacketSize) {}
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000093 std::string ToString() const;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000094
95 std::vector<uint32_t> ssrcs;
96
97 // Max RTP packet size delivered to send transport from VideoEngine.
98 size_t max_packet_size;
99
100 // RTP header extensions to use for this send stream.
101 std::vector<RtpExtension> extensions;
102
103 // See NackConfig for description.
104 NackConfig nack;
105
106 // See FecConfig for description.
107 FecConfig fec;
108
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000109 // Settings for RTP retransmission payload format, see RFC 4588 for
110 // details.
111 struct Rtx {
stefan@webrtc.org742386a2014-12-19 15:33:17 +0000112 Rtx() : payload_type(-1) {}
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +0000113 std::string ToString() const;
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000114 // SSRCs to use for the RTX streams.
115 std::vector<uint32_t> ssrcs;
116
117 // Payload type to use for the RTX stream.
118 int payload_type;
119 } rtx;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000120
121 // RTCP CNAME, see RFC 3550.
122 std::string c_name;
123 } rtp;
124
125 // Called for each I420 frame before encoding the frame. Can be used for
126 // effects, snapshots etc. 'NULL' disables the callback.
127 I420FrameCallback* pre_encode_callback;
128
129 // Called for each encoded frame, e.g. used for file storage. 'NULL'
130 // disables the callback.
sprang@webrtc.org40709352013-11-26 11:41:59 +0000131 EncodedFrameObserver* post_encode_callback;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000132
133 // Renderer for local preview. The local renderer will be called even if
134 // sending hasn't started. 'NULL' disables local rendering.
135 VideoRenderer* local_renderer;
136
137 // Expected delay needed by the renderer, i.e. the frame will be delivered
138 // this many milliseconds, if possible, earlier than expected render time.
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +0000139 // Only valid if |local_renderer| is set.
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000140 int render_delay_ms;
141
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000142 // Target delay in milliseconds. A positive value indicates this stream is
143 // used for streaming instead of a real-time call.
144 int target_delay_ms;
145
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000146 // True if the stream should be suspended when the available bitrate fall
147 // below the minimum configured bitrate. If this variable is false, the
148 // stream may send at a rate higher than the estimated available bitrate.
149 bool suspend_below_min_bitrate;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000150 };
151
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000152 // Gets interface used to insert captured frames. Valid as long as the
153 // VideoSendStream is valid.
154 virtual VideoSendStreamInput* Input() = 0;
155
pbos@webrtc.orga5c8d2c2014-04-24 11:13:21 +0000156 virtual void Start() = 0;
157 virtual void Stop() = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000158
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000159 // Set which streams to send. Must have at least as many SSRCs as configured
160 // in the config. Encoder settings are passed on to the encoder instance along
161 // with the VideoStream settings.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000162 virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000163
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000164 virtual Stats GetStats() = 0;
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000165
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000166 protected:
167 virtual ~VideoSendStream() {}
168};
169
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000170} // namespace webrtc
171
mflodman@webrtc.orgb429e512013-12-18 09:46:22 +0000172#endif // WEBRTC_VIDEO_SEND_STREAM_H_