Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(629)

Side by Side Diff: webrtc/config.h

Issue 3004723002: Move RtpExtension to api/ directory and config.h/.cc to call/. (Closed)
Patch Set: Fix nit. Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/call/video_send_stream.h ('k') | webrtc/config.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 // TODO(pbos): Move Config from common.h to here.
12
13 #ifndef WEBRTC_CONFIG_H_ 11 #ifndef WEBRTC_CONFIG_H_
14 #define WEBRTC_CONFIG_H_ 12 #define WEBRTC_CONFIG_H_
15 13
16 #include <string> 14 // TODO(holmer): Delete this file once downstream projects have been updated.
17 #include <vector>
18 15
19 #include "webrtc/common_types.h" 16 #include "webrtc/api/rtpparameters.h"
20 #include "webrtc/rtc_base/basictypes.h" 17 #include "webrtc/call/rtp_config.h"
21 #include "webrtc/rtc_base/optional.h"
22 #include "webrtc/rtc_base/refcount.h"
23 #include "webrtc/rtc_base/scoped_ref_ptr.h"
24 #include "webrtc/typedefs.h"
25
26 namespace webrtc {
27
28 // Settings for NACK, see RFC 4585 for details.
29 struct NackConfig {
30 NackConfig() : rtp_history_ms(0) {}
31 std::string ToString() const;
32 // Send side: the time RTP packets are stored for retransmissions.
33 // Receive side: the time the receiver is prepared to wait for
34 // retransmissions.
35 // Set to '0' to disable.
36 int rtp_history_ms;
37 };
38
39 // Settings for ULPFEC forward error correction.
40 // Set the payload types to '-1' to disable.
41 struct UlpfecConfig {
42 UlpfecConfig()
43 : ulpfec_payload_type(-1),
44 red_payload_type(-1),
45 red_rtx_payload_type(-1) {}
46 std::string ToString() const;
47 bool operator==(const UlpfecConfig& other) const;
48
49 // Payload type used for ULPFEC packets.
50 int ulpfec_payload_type;
51
52 // Payload type used for RED packets.
53 int red_payload_type;
54
55 // RTX payload type for RED payload.
56 int red_rtx_payload_type;
57 };
58
59 // RTP header extension, see RFC 5285.
60 struct RtpExtension {
61 RtpExtension() {}
62 RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
63 RtpExtension(const std::string& uri, int id, bool encrypt) : uri(uri),
64 id(id), encrypt(encrypt) {}
65 std::string ToString() const;
66 bool operator==(const RtpExtension& rhs) const {
67 return uri == rhs.uri && id == rhs.id && encrypt == rhs.encrypt;
68 }
69 static bool IsSupportedForAudio(const std::string& uri);
70 static bool IsSupportedForVideo(const std::string& uri);
71 // Return "true" if the given RTP header extension URI may be encrypted.
72 static bool IsEncryptionSupported(const std::string& uri);
73
74 // Returns the named header extension if found among all extensions,
75 // nullptr otherwise.
76 static const RtpExtension* FindHeaderExtensionByUri(
77 const std::vector<RtpExtension>& extensions,
78 const std::string& uri);
79
80 // Return a list of RTP header extensions with the non-encrypted extensions
81 // removed if both the encrypted and non-encrypted extension is present for
82 // the same URI.
83 static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
84 const std::vector<RtpExtension>& extensions);
85
86 // Header extension for audio levels, as defined in:
87 // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
88 static const char kAudioLevelUri[];
89 static const int kAudioLevelDefaultId;
90
91 // Header extension for RTP timestamp offset, see RFC 5450 for details:
92 // http://tools.ietf.org/html/rfc5450
93 static const char kTimestampOffsetUri[];
94 static const int kTimestampOffsetDefaultId;
95
96 // Header extension for absolute send time, see url for details:
97 // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
98 static const char kAbsSendTimeUri[];
99 static const int kAbsSendTimeDefaultId;
100
101 // Header extension for coordination of video orientation, see url for
102 // details:
103 // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126 114v120700p.pdf
104 static const char kVideoRotationUri[];
105 static const int kVideoRotationDefaultId;
106
107 // Header extension for video content type. E.g. default or screenshare.
108 static const char kVideoContentTypeUri[];
109 static const int kVideoContentTypeDefaultId;
110
111 // Header extension for video timing.
112 static const char kVideoTimingUri[];
113 static const int kVideoTimingDefaultId;
114
115 // Header extension for transport sequence number, see url for details:
116 // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
117 static const char kTransportSequenceNumberUri[];
118 static const int kTransportSequenceNumberDefaultId;
119
120 static const char kPlayoutDelayUri[];
121 static const int kPlayoutDelayDefaultId;
122
123 // Encryption of Header Extensions, see RFC 6904 for details:
124 // https://tools.ietf.org/html/rfc6904
125 static const char kEncryptHeaderExtensionsUri[];
126
127 // Inclusive min and max IDs for one-byte header extensions, per RFC5285.
128 static const int kMinId;
129 static const int kMaxId;
130
131 std::string uri;
132 int id = 0;
133 bool encrypt = false;
134 };
135
136 struct VideoStream {
137 VideoStream();
138 ~VideoStream();
139 std::string ToString() const;
140
141 size_t width;
142 size_t height;
143 int max_framerate;
144
145 int min_bitrate_bps;
146 int target_bitrate_bps;
147 int max_bitrate_bps;
148
149 int max_qp;
150
151 // Bitrate thresholds for enabling additional temporal layers. Since these are
152 // thresholds in between layers, we have one additional layer. One threshold
153 // gives two temporal layers, one below the threshold and one above, two give
154 // three, and so on.
155 // The VideoEncoder may redistribute bitrates over the temporal layers so a
156 // bitrate threshold of 100k and an estimate of 105k does not imply that we
157 // get 100k in one temporal layer and 5k in the other, just that the bitrate
158 // in the first temporal layer should not exceed 100k.
159 // TODO(kthelgason): Apart from a special case for two-layer screencast these
160 // thresholds are not propagated to the VideoEncoder. To be implemented.
161 std::vector<int> temporal_layer_thresholds_bps;
162 };
163
164 class VideoEncoderConfig {
165 public:
166 // These are reference counted to permit copying VideoEncoderConfig and be
167 // kept alive until all encoder_specific_settings go out of scope.
168 // TODO(kthelgason): Consider removing the need for copying VideoEncoderConfig
169 // and use rtc::Optional for encoder_specific_settings instead.
170 class EncoderSpecificSettings : public rtc::RefCountInterface {
171 public:
172 // TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
173 // not in use and encoder implementations ask for codec-specific structs
174 // directly.
175 void FillEncoderSpecificSettings(VideoCodec* codec_struct) const;
176
177 virtual void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const;
178 virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
179 virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
180 private:
181 ~EncoderSpecificSettings() override {}
182 friend class VideoEncoderConfig;
183 };
184
185 class H264EncoderSpecificSettings : public EncoderSpecificSettings {
186 public:
187 explicit H264EncoderSpecificSettings(const VideoCodecH264& specifics);
188 void FillVideoCodecH264(VideoCodecH264* h264_settings) const override;
189
190 private:
191 VideoCodecH264 specifics_;
192 };
193
194 class Vp8EncoderSpecificSettings : public EncoderSpecificSettings {
195 public:
196 explicit Vp8EncoderSpecificSettings(const VideoCodecVP8& specifics);
197 void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const override;
198
199 private:
200 VideoCodecVP8 specifics_;
201 };
202
203 class Vp9EncoderSpecificSettings : public EncoderSpecificSettings {
204 public:
205 explicit Vp9EncoderSpecificSettings(const VideoCodecVP9& specifics);
206 void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const override;
207
208 private:
209 VideoCodecVP9 specifics_;
210 };
211
212 enum class ContentType {
213 kRealtimeVideo,
214 kScreen,
215 };
216
217 class VideoStreamFactoryInterface : public rtc::RefCountInterface {
218 public:
219 // An implementation should return a std::vector<VideoStream> with the
220 // wanted VideoStream settings for the given video resolution.
221 // The size of the vector may not be larger than
222 // |encoder_config.number_of_streams|.
223 virtual std::vector<VideoStream> CreateEncoderStreams(
224 int width,
225 int height,
226 const VideoEncoderConfig& encoder_config) = 0;
227
228 protected:
229 ~VideoStreamFactoryInterface() override {}
230 };
231
232 VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
233 VideoEncoderConfig& operator=(const VideoEncoderConfig&) = delete;
234
235 // Mostly used by tests. Avoid creating copies if you can.
236 VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
237
238 VideoEncoderConfig();
239 VideoEncoderConfig(VideoEncoderConfig&&);
240 ~VideoEncoderConfig();
241 std::string ToString() const;
242
243 rtc::scoped_refptr<VideoStreamFactoryInterface> video_stream_factory;
244 std::vector<SpatialLayer> spatial_layers;
245 ContentType content_type;
246 rtc::scoped_refptr<const EncoderSpecificSettings> encoder_specific_settings;
247
248 // Padding will be used up to this bitrate regardless of the bitrate produced
249 // by the encoder. Padding above what's actually produced by the encoder helps
250 // maintaining a higher bitrate estimate. Padding will however not be sent
251 // unless the estimated bandwidth indicates that the link can handle it.
252 int min_transmit_bitrate_bps;
253 int max_bitrate_bps;
254
255 // Max number of encoded VideoStreams to produce.
256 size_t number_of_streams;
257
258 private:
259 // Access to the copy constructor is private to force use of the Copy()
260 // method for those exceptional cases where we do use it.
261 VideoEncoderConfig(const VideoEncoderConfig&);
262 };
263
264 } // namespace webrtc
265 18
266 #endif // WEBRTC_CONFIG_H_ 19 #endif // WEBRTC_CONFIG_H_
OLDNEW
« no previous file with comments | « webrtc/call/video_send_stream.h ('k') | webrtc/config.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698