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mflodman@webrtc.org06e80262013-04-18 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.org5e0cbcf2013-12-18 09:46:22 +000011#ifndef WEBRTC_VIDEO_RECEIVE_STREAM_H_
12#define WEBRTC_VIDEO_RECEIVE_STREAM_H_
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000013
pbos@webrtc.org5b080522014-01-20 14:43:55 +000014#include <map>
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000015#include <string>
16#include <vector>
17
18#include "webrtc/common_types.h"
pbos@webrtc.org24e20892013-10-28 16:32:01 +000019#include "webrtc/config.h"
20#include "webrtc/frame_callback.h"
21#include "webrtc/transport.h"
22#include "webrtc/video_renderer.h"
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000023
24namespace webrtc {
25
pbos@webrtc.org51e01012013-10-17 14:14:42 +000026namespace newapi {
27// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
28// RTCP mode is described by RFC 5506.
pbos@webrtc.orgc71929d2014-01-24 09:30:53 +000029enum RtcpMode { kRtcpCompound, kRtcpReducedSize };
pbos@webrtc.org51e01012013-10-17 14:14:42 +000030} // namespace newapi
31
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000032class VideoDecoder;
33
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000034// TODO(mflodman) Move all these settings to VideoDecoder and move the
35// declaration to common_types.h.
36struct ExternalVideoDecoder {
pbos@webrtc.orgb2d1a402013-05-28 08:04:45 +000037 ExternalVideoDecoder()
38 : decoder(NULL), payload_type(0), renderer(false), expected_delay_ms(0) {}
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000039 // The actual decoder.
40 VideoDecoder* decoder;
41
42 // Received RTP packets with this payload type will be sent to this decoder
43 // instance.
44 int payload_type;
45
46 // 'true' if the decoder handles rendering as well.
47 bool renderer;
48
49 // The expected delay for decoding and rendering, i.e. the frame will be
50 // delivered this many milliseconds, if possible, earlier than the ideal
51 // render time.
52 // Note: Ignored if 'renderer' is false.
53 int expected_delay_ms;
54};
55
mflodman@webrtc.org06e80262013-04-18 12:02:52 +000056class VideoReceiveStream {
57 public:
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +000058 struct Stats {
59 Stats()
60 : network_frame_rate(0),
61 decode_frame_rate(0),
62 render_frame_rate(0),
63 key_frames(0),
64 delta_frames(0),
65 video_packets(0),
66 retransmitted_packets(0),
67 fec_packets(0),
68 padding_packets(0),
69 discarded_packets(0),
70 received_bitrate_bps(0),
71 receive_side_delay_ms(0) {}
72 RtpStatistics rtp_stats;
73 int network_frame_rate;
74 int decode_frame_rate;
75 int render_frame_rate;
76 uint32_t key_frames;
77 uint32_t delta_frames;
78 uint32_t video_packets;
79 uint32_t retransmitted_packets;
80 uint32_t fec_packets;
81 uint32_t padding_packets;
82 uint32_t discarded_packets;
83 int32_t received_bitrate_bps;
84 int receive_side_delay_ms;
85 };
86
87 class StatsCallback {
88 public:
89 virtual ~StatsCallback() {}
90 virtual void ReceiveStats(const Stats& stats) = 0;
91 };
92
93 struct Config {
94 Config()
95 : renderer(NULL),
96 render_delay_ms(0),
97 audio_channel_id(0),
98 pre_decode_callback(NULL),
pbos@webrtc.org63301bd2013-10-21 10:34:43 +000099 pre_render_callback(NULL),
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +0000100 target_delay_ms(0) {}
pbos@webrtc.orgce851092013-08-05 12:01:36 +0000101 // Codecs the receive stream can receive.
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +0000102 std::vector<VideoCodec> codecs;
103
104 // Receive-stream specific RTP settings.
105 struct Rtp {
pbos@webrtc.org4b50db12013-12-03 10:13:04 +0000106 Rtp()
107 : remote_ssrc(0),
108 local_ssrc(0),
mflodman@webrtc.org7ff40892013-12-13 16:36:28 +0000109 rtcp_mode(newapi::kRtcpReducedSize),
110 remb(false) {}
pbos@webrtc.org51e01012013-10-17 14:14:42 +0000111
pbos@webrtc.org4b50db12013-12-03 10:13:04 +0000112 // Synchronization source (stream identifier) to be received.
113 uint32_t remote_ssrc;
114 // Sender SSRC used for sending RTCP (such as receiver reports).
115 uint32_t local_ssrc;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +0000116
pbos@webrtc.org51e01012013-10-17 14:14:42 +0000117 // See RtcpMode for description.
118 newapi::RtcpMode rtcp_mode;
119
asapersson@webrtc.orgb4263e02014-01-20 08:34:49 +0000120 // Extended RTCP settings.
121 struct RtcpXr {
122 RtcpXr() : receiver_reference_time_report(false) {}
123
124 // True if RTCP Receiver Reference Time Report Block extension
125 // (RFC 3611) should be enabled.
126 bool receiver_reference_time_report;
127 } rtcp_xr;
128
mflodman@webrtc.org7ff40892013-12-13 16:36:28 +0000129 // See draft-alvestrand-rmcat-remb for information.
130 bool remb;
131
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +0000132 // See NackConfig for description.
133 NackConfig nack;
134
135 // See FecConfig for description.
136 FecConfig fec;
137
pbos@webrtc.orgc71929d2014-01-24 09:30:53 +0000138 // RTX settings for incoming video payloads that may be received. RTX is
139 // disabled if there's no config present.
140 struct Rtx {
141 Rtx() : ssrc(0), payload_type(0) {}
142
143 // SSRCs to use for the RTX streams.
144 uint32_t ssrc;
145
146 // Payload type to use for the RTX stream.
147 int payload_type;
148 };
149
150 // Map from video RTP payload type -> RTX config.
151 typedef std::map<int, Rtx> RtxMap;
152 RtxMap rtx;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +0000153
154 // RTP header extensions used for the received stream.
155 std::vector<RtpExtension> extensions;
156 } rtp;
157
158 // VideoRenderer will be called for each decoded frame. 'NULL' disables
159 // rendering of this stream.
160 VideoRenderer* renderer;
161
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.
165 int render_delay_ms;
166
167 // Audio channel corresponding to this video stream, used for audio/video
168 // synchronization. 'audio_channel_id' is ignored if no VoiceEngine is set
169 // when creating the VideoEngine instance. '-1' disables a/v sync.
170 int audio_channel_id;
171
172 // Called for each incoming video frame, i.e. in encoded state. E.g. used
173 // when
174 // saving the stream to a file. 'NULL' disables the callback.
175 EncodedFrameObserver* pre_decode_callback;
176
177 // Called for each decoded frame. E.g. used when adding effects to the
178 // decoded
179 // stream. 'NULL' disables the callback.
pbos@webrtc.org63301bd2013-10-21 10:34:43 +0000180 I420FrameCallback* pre_render_callback;
pbos@webrtc.org6f1c3ef2013-06-05 11:33:21 +0000181
182 // External video decoders to be used if incoming payload type matches the
183 // registered type for an external decoder.
184 std::vector<ExternalVideoDecoder> external_decoders;
185
186 // Target delay in milliseconds. A positive value indicates this stream is
187 // used for streaming instead of a real-time call.
188 int target_delay_ms;
189
190 // Callback for periodically receiving receiver stats.
191 StatsCallback* stats_callback;
192 };
193
pbos@webrtc.org7f9f8402013-11-20 11:36:47 +0000194 virtual void StartReceiving() = 0;
195 virtual void StopReceiving() = 0;
mflodman@webrtc.org06e80262013-04-18 12:02:52 +0000196
197 // TODO(mflodman) Replace this with callback.
198 virtual void GetCurrentReceiveCodec(VideoCodec* receive_codec) = 0;
199
mflodman@webrtc.org06e80262013-04-18 12:02:52 +0000200 protected:
201 virtual ~VideoReceiveStream() {}
202};
203
mflodman@webrtc.org06e80262013-04-18 12:02:52 +0000204} // namespace webrtc
205
mflodman@webrtc.org5e0cbcf2013-12-18 09:46:22 +0000206#endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_