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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_RECEIVE_STREAM_H_
12#define WEBRTC_VIDEO_RECEIVE_STREAM_H_
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000013
asaperssonf8cdd182016-03-15 01:00:47 -070014#include <limits>
pbos@webrtc.orge02d4752014-01-20 14:43:55 +000015#include <map>
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000016#include <string>
17#include <vector>
18
19#include "webrtc/common_types.h"
pbosa96b60b2016-04-18 21:12:48 -070020#include "webrtc/common_video/include/frame_callback.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000021#include "webrtc/config.h"
pbosa96b60b2016-04-18 21:12:48 -070022#include "webrtc/media/base/videosinkinterface.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000023#include "webrtc/transport.h"
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000024
25namespace webrtc {
26
27class VideoDecoder;
28
pbos1ba8d392016-05-01 20:18:34 -070029class VideoReceiveStream {
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000030 public:
pbos@webrtc.org776e6f22014-10-29 15:28:39 +000031 // TODO(mflodman) Move all these settings to VideoDecoder and move the
32 // declaration to common_types.h.
33 struct Decoder {
pbos@webrtc.org32e85282015-01-15 10:09:39 +000034 std::string ToString() const;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +000035
36 // The actual decoder instance.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +020037 VideoDecoder* decoder = nullptr;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +000038
39 // Received RTP packets with this payload type will be sent to this decoder
40 // instance.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +020041 int payload_type = 0;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +000042
43 // Name of the decoded payload (such as VP8). Maps back to the depacketizer
44 // used to unpack incoming packets.
45 std::string payload_name;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +000046 };
47
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000048 struct Stats {
49 int network_frame_rate = 0;
50 int decode_frame_rate = 0;
51 int render_frame_rate = 0;
sprang@webrtc.org09315702014-02-07 12:06:29 +000052
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000053 // Decoder stats.
Peter Boströmb7d9a972015-12-18 16:01:11 +010054 std::string decoder_implementation_name = "unknown";
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000055 FrameCounts frame_counts;
56 int decode_ms = 0;
57 int max_decode_ms = 0;
58 int current_delay_ms = 0;
59 int target_delay_ms = 0;
60 int jitter_buffer_ms = 0;
61 int min_playout_delay_ms = 0;
Peter Boströmc4188fd2015-04-24 15:16:03 +020062 int render_delay_ms = 10;
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000063
pbosf42376c2015-08-28 07:35:32 -070064 int current_payload_type = -1;
65
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000066 int total_bitrate_bps = 0;
67 int discarded_packets = 0;
68
asaperssonf8cdd182016-03-15 01:00:47 -070069 int sync_offset_ms = std::numeric_limits<int>::max();
70
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000071 uint32_t ssrc = 0;
sprang@webrtc.org09315702014-02-07 12:06:29 +000072 std::string c_name;
pbos@webrtc.org09c77b92015-02-25 10:42:16 +000073 StreamDataCounters rtp_stats;
74 RtcpPacketTypeCounter rtcp_packet_type_counts;
75 RtcpStatistics rtcp_stats;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +000076 };
77
78 struct Config {
Tommi733b5472016-06-10 17:58:01 +020079 private:
80 // Access to the copy constructor is private to force use of the Copy()
81 // method for those exceptional cases where we do use it.
82 Config(const Config&) = default;
83
84 public:
solenberg4fbae2b2015-08-28 04:07:10 -070085 Config() = delete;
Tommi733b5472016-06-10 17:58:01 +020086 Config(Config&&) = default;
pbos2d566682015-09-28 09:59:31 -070087 explicit Config(Transport* rtcp_send_transport)
solenberg4fbae2b2015-08-28 04:07:10 -070088 : rtcp_send_transport(rtcp_send_transport) {}
89
Tommi733b5472016-06-10 17:58:01 +020090 Config& operator=(Config&&) = default;
91 Config& operator=(const Config&) = delete;
92
93 // Mostly used by tests. Avoid creating copies if you can.
94 Config Copy() const { return Config(*this); }
95
pbos@webrtc.org32e85282015-01-15 10:09:39 +000096 std::string ToString() const;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +000097
98 // Decoders for every payload that we can receive.
99 std::vector<Decoder> decoders;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000100
101 // Receive-stream specific RTP settings.
102 struct Rtp {
pbos@webrtc.org32e85282015-01-15 10:09:39 +0000103 std::string ToString() const;
pbos@webrtc.orgc11148b2013-10-17 14:14:42 +0000104
pbos@webrtc.orgb613b5a2013-12-03 10:13:04 +0000105 // Synchronization source (stream identifier) to be received.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200106 uint32_t remote_ssrc = 0;
pbos@webrtc.orgb613b5a2013-12-03 10:13:04 +0000107 // Sender SSRC used for sending RTCP (such as receiver reports).
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200108 uint32_t local_ssrc = 0;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000109
pbos@webrtc.orgc11148b2013-10-17 14:14:42 +0000110 // See RtcpMode for description.
pbosda903ea2015-10-02 02:36:56 -0700111 RtcpMode rtcp_mode = RtcpMode::kCompound;
pbos@webrtc.orgc11148b2013-10-17 14:14:42 +0000112
asapersson@webrtc.orgefaeda02014-01-20 08:34:49 +0000113 // Extended RTCP settings.
114 struct RtcpXr {
asapersson@webrtc.orgefaeda02014-01-20 08:34:49 +0000115 // True if RTCP Receiver Reference Time Report Block extension
116 // (RFC 3611) should be enabled.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200117 bool receiver_reference_time_report = false;
asapersson@webrtc.orgefaeda02014-01-20 08:34:49 +0000118 } rtcp_xr;
119
mflodman@webrtc.org92c27932013-12-13 16:36:28 +0000120 // See draft-alvestrand-rmcat-remb for information.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200121 bool remb = false;
mflodman@webrtc.org92c27932013-12-13 16:36:28 +0000122
stefan43edf0f2015-11-20 18:05:48 -0800123 // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
124 bool transport_cc = false;
125
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000126 // See NackConfig for description.
127 NackConfig nack;
128
129 // See FecConfig for description.
130 FecConfig fec;
131
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000132 // RTX settings for incoming video payloads that may be received. RTX is
133 // disabled if there's no config present.
134 struct Rtx {
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000135 // SSRCs to use for the RTX streams.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200136 uint32_t ssrc = 0;
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000137
138 // Payload type to use for the RTX stream.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200139 int payload_type = 0;
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000140 };
141
142 // Map from video RTP payload type -> RTX config.
143 typedef std::map<int, Rtx> RtxMap;
144 RtxMap rtx;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000145
noahric65220a72015-10-14 11:29:49 -0700146 // If set to true, the RTX payload type mapping supplied in |rtx| will be
147 // used when restoring RTX packets. Without it, RTX packets will always be
148 // restored to the last non-RTX packet payload type received.
149 bool use_rtx_payload_mapping_on_restore = false;
150
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000151 // RTP header extensions used for the received stream.
152 std::vector<RtpExtension> extensions;
153 } rtp;
154
solenberg4fbae2b2015-08-28 04:07:10 -0700155 // Transport for outgoing packets (RTCP).
pbos2d566682015-09-28 09:59:31 -0700156 Transport* rtcp_send_transport = nullptr;
solenberg4fbae2b2015-08-28 04:07:10 -0700157
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200158 // VideoRenderer will be called for each decoded frame. 'nullptr' disables
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000159 // rendering of this stream.
nisse7ade7b32016-03-23 04:48:10 -0700160 rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000161
162 // Expected delay needed by the renderer, i.e. the frame will be delivered
163 // this many milliseconds, if possible, earlier than the ideal render time.
164 // Only valid if 'renderer' is set.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200165 int render_delay_ms = 10;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000166
nisse7ade7b32016-03-23 04:48:10 -0700167 // If set, pass frames on to the renderer as soon as they are
168 // available.
169 bool disable_prerenderer_smoothing = false;
170
pbos8fc7fa72015-07-15 08:02:58 -0700171 // Identifier for an A/V synchronization group. Empty string to disable.
172 // TODO(pbos): Synchronize streams in a sync group, not just video streams
173 // to one of the audio streams.
174 std::string sync_group;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000175
176 // Called for each incoming video frame, i.e. in encoded state. E.g. used
177 // when
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200178 // saving the stream to a file. 'nullptr' disables the callback.
179 EncodedFrameObserver* pre_decode_callback = nullptr;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000180
181 // Called for each decoded frame. E.g. used when adding effects to the
182 // decoded
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200183 // stream. 'nullptr' disables the callback.
Tommibd3380f2016-06-10 17:38:17 +0200184 // TODO(tommi): This seems to be only used by a test or two. Consider
185 // removing it (and use an appropriate alternative in the tests) as well
186 // as the associated code in VideoStreamDecoder.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200187 I420FrameCallback* pre_render_callback = nullptr;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000188
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000189 // Target delay in milliseconds. A positive value indicates this stream is
190 // used for streaming instead of a real-time call.
Fredrik Solenberg78fb3b32015-06-11 12:38:38 +0200191 int target_delay_ms = 0;
pbos@webrtc.org025f4f12013-06-05 11:33:21 +0000192 };
193
pbos1ba8d392016-05-01 20:18:34 -0700194 // Starts stream activity.
195 // When a stream is active, it can receive, process and deliver packets.
196 virtual void Start() = 0;
197 // Stops stream activity.
198 // When a stream is stopped, it can't receive, process or deliver packets.
199 virtual void Stop() = 0;
200
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000201 // TODO(pbos): Add info on currently-received codec to Stats.
202 virtual Stats GetStats() const = 0;
pbos1ba8d392016-05-01 20:18:34 -0700203
204 protected:
205 virtual ~VideoReceiveStream() {}
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000206};
207
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000208} // namespace webrtc
209
mflodman@webrtc.orgb429e512013-12-18 09:46:22 +0000210#endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_