Line data Source code
1 : /*
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 :
11 : #ifndef WEBRTC_VIDEO_SEND_STREAM_H_
12 : #define WEBRTC_VIDEO_SEND_STREAM_H_
13 :
14 : #include <map>
15 : #include <string>
16 : #include <utility>
17 : #include <vector>
18 : #include <utility>
19 :
20 : #include "webrtc/api/call/transport.h"
21 : #include "webrtc/base/platform_file.h"
22 : #include "webrtc/common_types.h"
23 : #include "webrtc/common_video/include/frame_callback.h"
24 : #include "webrtc/config.h"
25 : #include "webrtc/media/base/videosinkinterface.h"
26 : #include "webrtc/media/base/videosourceinterface.h"
27 :
28 : namespace webrtc {
29 :
30 : class VideoEncoder;
31 :
32 0 : class VideoSendStream {
33 : public:
34 0 : struct StreamStats {
35 : std::string ToString() const;
36 :
37 : FrameCounts frame_counts;
38 : bool is_rtx = false;
39 : bool is_flexfec = false;
40 : int width = 0;
41 : int height = 0;
42 : // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
43 : int total_bitrate_bps = 0;
44 : int retransmit_bitrate_bps = 0;
45 : int avg_delay_ms = 0;
46 : int max_delay_ms = 0;
47 : StreamDataCounters rtp_stats;
48 : RtcpPacketTypeCounter rtcp_packet_type_counts;
49 : RtcpStatistics rtcp_stats;
50 : };
51 :
52 0 : struct Stats {
53 : std::string ToString(int64_t time_ms) const;
54 : std::string encoder_implementation_name = "unknown";
55 : int input_frame_rate = 0;
56 : int encode_frame_rate = 0;
57 : int avg_encode_time_ms = 0;
58 : int encode_usage_percent = 0;
59 : uint32_t frames_encoded = 0;
60 : rtc::Optional<uint64_t> qp_sum;
61 : // Bitrate the encoder is currently configured to use due to bandwidth
62 : // limitations.
63 : int target_media_bitrate_bps = 0;
64 : // Bitrate the encoder is actually producing.
65 : int media_bitrate_bps = 0;
66 : // Media bitrate this VideoSendStream is configured to prefer if there are
67 : // no bandwidth limitations.
68 : int preferred_media_bitrate_bps = 0;
69 : bool suspended = false;
70 : bool bw_limited_resolution = false;
71 : bool cpu_limited_resolution = false;
72 : // Total number of times resolution as been requested to be changed due to
73 : // CPU adaptation.
74 : int number_of_cpu_adapt_changes = 0;
75 : std::map<uint32_t, StreamStats> substreams;
76 : };
77 :
78 0 : struct Config {
79 : public:
80 : Config() = delete;
81 0 : Config(Config&&) = default;
82 0 : explicit Config(Transport* send_transport)
83 0 : : send_transport(send_transport) {}
84 :
85 : Config& operator=(Config&&) = default;
86 : Config& operator=(const Config&) = delete;
87 :
88 : // Mostly used by tests. Avoid creating copies if you can.
89 0 : Config Copy() const { return Config(*this); }
90 :
91 : std::string ToString() const;
92 :
93 0 : struct EncoderSettings {
94 0 : EncoderSettings() = default;
95 : EncoderSettings(std::string payload_name,
96 : int payload_type,
97 : VideoEncoder* encoder)
98 : : payload_name(std::move(payload_name)),
99 : payload_type(payload_type),
100 : encoder(encoder) {}
101 : std::string ToString() const;
102 :
103 : std::string payload_name;
104 : int payload_type = -1;
105 :
106 : // TODO(sophiechang): Delete this field when no one is using internal
107 : // sources anymore.
108 : bool internal_source = false;
109 :
110 : // Allow 100% encoder utilization. Used for HW encoders where CPU isn't
111 : // expected to be the limiting factor, but a chip could be running at
112 : // 30fps (for example) exactly.
113 : bool full_overuse_time = false;
114 :
115 : // Uninitialized VideoEncoder instance to be used for encoding. Will be
116 : // initialized from inside the VideoSendStream.
117 : VideoEncoder* encoder = nullptr;
118 : } encoder_settings;
119 :
120 : static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
121 0 : struct Rtp {
122 : std::string ToString() const;
123 :
124 : std::vector<uint32_t> ssrcs;
125 :
126 : // See RtcpMode for description.
127 : RtcpMode rtcp_mode = RtcpMode::kCompound;
128 :
129 : // Max RTP packet size delivered to send transport from VideoEngine.
130 : size_t max_packet_size = kDefaultMaxPacketSize;
131 :
132 : // RTP header extensions to use for this send stream.
133 : std::vector<RtpExtension> extensions;
134 :
135 : // See NackConfig for description.
136 : NackConfig nack;
137 :
138 : // See UlpfecConfig for description.
139 : UlpfecConfig ulpfec;
140 :
141 0 : struct Flexfec {
142 : // Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
143 : int payload_type = -1;
144 :
145 : // SSRC of FlexFEC stream.
146 : uint32_t ssrc = 0;
147 :
148 : // Vector containing a single element, corresponding to the SSRC of the
149 : // media stream being protected by this FlexFEC stream.
150 : // The vector MUST have size 1.
151 : //
152 : // TODO(brandtr): Update comment above when we support
153 : // multistream protection.
154 : std::vector<uint32_t> protected_media_ssrcs;
155 : } flexfec;
156 :
157 : // Settings for RTP retransmission payload format, see RFC 4588 for
158 : // details.
159 0 : struct Rtx {
160 : std::string ToString() const;
161 : // SSRCs to use for the RTX streams.
162 : std::vector<uint32_t> ssrcs;
163 :
164 : // Payload type to use for the RTX stream.
165 : int payload_type = -1;
166 : } rtx;
167 :
168 : // RTCP CNAME, see RFC 3550.
169 : std::string c_name;
170 :
171 : std::vector<std::string> rids;
172 : } rtp;
173 :
174 : // Transport for outgoing packets.
175 : Transport* send_transport = nullptr;
176 :
177 : // Called for each I420 frame before encoding the frame. Can be used for
178 : // effects, snapshots etc. 'nullptr' disables the callback.
179 : rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
180 :
181 : // Called for each encoded frame, e.g. used for file storage. 'nullptr'
182 : // disables the callback. Also measures timing and passes the time
183 : // spent on encoding. This timing will not fire if encoding takes longer
184 : // than the measuring window, since the sample data will have been dropped.
185 : EncodedFrameObserver* post_encode_callback = nullptr;
186 :
187 : // Expected delay needed by the renderer, i.e. the frame will be delivered
188 : // this many milliseconds, if possible, earlier than expected render time.
189 : // Only valid if |local_renderer| is set.
190 : int render_delay_ms = 0;
191 :
192 : // Target delay in milliseconds. A positive value indicates this stream is
193 : // used for streaming instead of a real-time call.
194 : int target_delay_ms = 0;
195 :
196 : // True if the stream should be suspended when the available bitrate fall
197 : // below the minimum configured bitrate. If this variable is false, the
198 : // stream may send at a rate higher than the estimated available bitrate.
199 : bool suspend_below_min_bitrate = false;
200 :
201 : // Enables periodic bandwidth probing in application-limited region.
202 : bool periodic_alr_bandwidth_probing = false;
203 :
204 : private:
205 : // Access to the copy constructor is private to force use of the Copy()
206 : // method for those exceptional cases where we do use it.
207 0 : Config(const Config&) = default;
208 : };
209 :
210 : // Starts stream activity.
211 : // When a stream is active, it can receive, process and deliver packets.
212 : virtual void Start() = 0;
213 : // Stops stream activity.
214 : // When a stream is stopped, it can't receive, process or deliver packets.
215 : virtual void Stop() = 0;
216 :
217 : // Based on the spec in
218 : // https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
219 : enum class DegradationPreference {
220 : kMaintainResolution,
221 : // TODO(perkj): Implement kMaintainFrameRate. kBalanced will drop frames
222 : // if the encoder overshoots or the encoder can not encode fast enough.
223 : kBalanced,
224 : };
225 : virtual void SetSource(
226 : rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
227 : const DegradationPreference& degradation_preference) = 0;
228 :
229 : // Gets interface used to signal the current CPU work level to the encoder.
230 : // Valid as long as the VideoSendStream is valid.
231 : virtual CPULoadStateObserver* LoadStateObserver() = 0;
232 :
233 : // Set which streams to send. Must have at least as many SSRCs as configured
234 : // in the config. Encoder settings are passed on to the encoder instance along
235 : // with the VideoStream settings.
236 : virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
237 :
238 : virtual Stats GetStats() = 0;
239 :
240 : // Takes ownership of each file, is responsible for closing them later.
241 : // Calling this method will close and finalize any current logs.
242 : // Some codecs produce multiple streams (VP8 only at present), each of these
243 : // streams will log to a separate file. kMaxSimulcastStreams in common_types.h
244 : // gives the max number of such streams. If there is no file for a stream, or
245 : // the file is rtc::kInvalidPlatformFileValue, frames from that stream will
246 : // not be logged.
247 : // If a frame to be written would make the log too large the write fails and
248 : // the log is closed and finalized. A |byte_limit| of 0 means no limit.
249 : virtual void EnableEncodedFrameRecording(
250 : const std::vector<rtc::PlatformFile>& files,
251 : size_t byte_limit) = 0;
252 : inline void DisableEncodedFrameRecording() {
253 : EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
254 : }
255 :
256 : protected:
257 0 : virtual ~VideoSendStream() {}
258 : };
259 :
260 : } // namespace webrtc
261 :
262 : #endif // WEBRTC_VIDEO_SEND_STREAM_H_
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