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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"
Jelena Marusiccd670222015-07-16 09:30:09 +020020#include "webrtc/stream.h"
solenberg4fbae2b2015-08-28 04:07:10 -070021#include "webrtc/transport.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000022#include "webrtc/video_renderer.h"
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000023
24namespace webrtc {
25
26class VideoEncoder;
27
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000028// Class to deliver captured frame to the video send stream.
Peter Boström4b91bd02015-06-26 06:58:16 +020029class VideoCaptureInput {
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000030 public:
pbos@webrtc.org724947b2013-12-11 16:26:16 +000031 // These methods do not lock internally and must be called sequentially.
32 // If your application switches input sources synchronization must be done
33 // externally to make sure that any old frames are not delivered concurrently.
Miguel Casas-Sanchez47650702015-05-29 17:21:40 -070034 virtual void IncomingCapturedFrame(const VideoFrame& video_frame) = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000035
36 protected:
Peter Boström4b91bd02015-06-26 06:58:16 +020037 virtual ~VideoCaptureInput() {}
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000038};
39
Jelena Marusiccd670222015-07-16 09:30:09 +020040class VideoSendStream : public SendStream {
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000041 public:
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000042 struct StreamStats {
43 FrameCounts frame_counts;
44 int width = 0;
45 int height = 0;
46 // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
47 int total_bitrate_bps = 0;
48 int retransmit_bitrate_bps = 0;
49 int avg_delay_ms = 0;
50 int max_delay_ms = 0;
51 StreamDataCounters rtp_stats;
52 RtcpPacketTypeCounter rtcp_packet_type_counts;
53 RtcpStatistics rtcp_stats;
54 };
55
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000056 struct Stats {
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +020057 int input_frame_rate = 0;
58 int encode_frame_rate = 0;
59 int avg_encode_time_ms = 0;
60 int encode_usage_percent = 0;
61 int target_media_bitrate_bps = 0;
62 int media_bitrate_bps = 0;
63 bool suspended = false;
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 {
solenberg4fbae2b2015-08-28 04:07:10 -070068 Config() = delete;
69 explicit Config(newapi::Transport* send_transport)
70 : send_transport(send_transport) {}
71
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000072 std::string ToString() const;
73
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000074 struct EncoderSettings {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000075 std::string ToString() const;
76
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000077 std::string payload_name;
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +020078 int payload_type = -1;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000079
sophiechang47d78cc2015-09-03 18:24:44 -070080 // TODO(sophiechang): Delete this field when no one is using internal
81 // sources anymore.
82 bool internal_source = false;
83
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000084 // Uninitialized VideoEncoder instance to be used for encoding. Will be
85 // initialized from inside the VideoSendStream.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +020086 VideoEncoder* encoder = nullptr;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000087 } encoder_settings;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000088
sprang@webrtc.org25fce9a2013-10-16 13:29:14 +000089 static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000090 struct Rtp {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000091 std::string ToString() const;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000092
93 std::vector<uint32_t> ssrcs;
94
95 // Max RTP packet size delivered to send transport from VideoEngine.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +020096 size_t max_packet_size = kDefaultMaxPacketSize;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000097
98 // RTP header extensions to use for this send stream.
99 std::vector<RtpExtension> extensions;
100
101 // See NackConfig for description.
102 NackConfig nack;
103
104 // See FecConfig for description.
105 FecConfig fec;
106
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000107 // Settings for RTP retransmission payload format, see RFC 4588 for
108 // details.
109 struct Rtx {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +0000110 std::string ToString() const;
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000111 // SSRCs to use for the RTX streams.
112 std::vector<uint32_t> ssrcs;
113
114 // Payload type to use for the RTX stream.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200115 int payload_type = -1;
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000116 } rtx;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000117
118 // RTCP CNAME, see RFC 3550.
119 std::string c_name;
120 } rtp;
121
solenberg4fbae2b2015-08-28 04:07:10 -0700122 // Transport for outgoing packets.
123 newapi::Transport* send_transport = nullptr;
124
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000125 // Called for each I420 frame before encoding the frame. Can be used for
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200126 // effects, snapshots etc. 'nullptr' disables the callback.
127 I420FrameCallback* pre_encode_callback = nullptr;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000128
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200129 // Called for each encoded frame, e.g. used for file storage. 'nullptr'
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000130 // disables the callback.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200131 EncodedFrameObserver* post_encode_callback = nullptr;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000132
133 // Renderer for local preview. The local renderer will be called even if
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200134 // sending hasn't started. 'nullptr' disables local rendering.
135 VideoRenderer* local_renderer = nullptr;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000136
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.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200140 int render_delay_ms = 0;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000141
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.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200144 int target_delay_ms = 0;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000145
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.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200149 bool suspend_below_min_bitrate = false;
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.
Peter Boström4b91bd02015-06-26 06:58:16 +0200154 virtual VideoCaptureInput* Input() = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000155
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000156 // Set which streams to send. Must have at least as many SSRCs as configured
157 // in the config. Encoder settings are passed on to the encoder instance along
158 // with the VideoStream settings.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000159 virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000160
pbos@webrtc.org273a4142014-12-01 15:23:21 +0000161 virtual Stats GetStats() = 0;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000162};
163
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000164} // namespace webrtc
165
mflodman@webrtc.orgb429e512013-12-18 09:46:22 +0000166#endif // WEBRTC_VIDEO_SEND_STREAM_H_