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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
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000011// TODO(pbos): Move Config from common.h to here.
12
pbos@webrtc.org3c107582014-07-20 15:27:35 +000013#ifndef WEBRTC_CONFIG_H_
14#define WEBRTC_CONFIG_H_
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000015
16#include <string>
pbos@webrtc.org5860de02013-09-16 13:01:47 +000017#include <vector>
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000018
sprang@webrtc.orgccd42842014-01-07 09:54:34 +000019#include "webrtc/common_types.h"
pbos@webrtc.orgce90eff2013-11-20 11:48:56 +000020#include "webrtc/typedefs.h"
21
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000022namespace webrtc {
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000023
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000024// Settings for NACK, see RFC 4585 for details.
25struct NackConfig {
pbos@webrtc.orgeceb5322013-05-28 08:04:45 +000026 NackConfig() : rtp_history_ms(0) {}
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000027 // Send side: the time RTP packets are stored for retransmissions.
28 // Receive side: the time the receiver is prepared to wait for
29 // retransmissions.
pbos@webrtc.orgeceb5322013-05-28 08:04:45 +000030 // Set to '0' to disable.
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000031 int rtp_history_ms;
32};
33
34// Settings for forward error correction, see RFC 5109 for details. Set the
35// payload types to '-1' to disable.
36struct FecConfig {
Shao Changbine62202f2015-04-21 20:24:50 +080037 FecConfig()
38 : ulpfec_payload_type(-1),
39 red_payload_type(-1),
40 red_rtx_payload_type(-1) {}
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000041 std::string ToString() const;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000042 // Payload type used for ULPFEC packets.
43 int ulpfec_payload_type;
44
45 // Payload type used for RED packets.
46 int red_payload_type;
Shao Changbine62202f2015-04-21 20:24:50 +080047
48 // RTX payload type for RED payload.
49 int red_rtx_payload_type;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000050};
51
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000052// RTP header extension to use for the video stream, see RFC 5285.
53struct RtpExtension {
pbos@webrtc.org3c107582014-07-20 15:27:35 +000054 RtpExtension(const std::string& name, int id) : name(name), id(id) {}
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000055 std::string ToString() const;
pbos@webrtc.org3c107582014-07-20 15:27:35 +000056 static bool IsSupported(const std::string& name);
57
pbos@webrtc.orgce90eff2013-11-20 11:48:56 +000058 static const char* kTOffset;
59 static const char* kAbsSendTime;
guoweis@webrtc.orgfdd10572015-03-12 20:50:57 +000060 static const char* kVideoRotation;
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +000061 std::string name;
62 int id;
63};
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000064
65struct VideoStream {
kwiberg@webrtc.orgac2d27d2015-02-26 13:59:22 +000066 VideoStream();
67 ~VideoStream();
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000068 std::string ToString() const;
69
70 size_t width;
71 size_t height;
72 int max_framerate;
73
74 int min_bitrate_bps;
75 int target_bitrate_bps;
76 int max_bitrate_bps;
77
78 int max_qp;
79
pbos@webrtc.orgb7ed7792014-10-31 13:08:10 +000080 // Bitrate thresholds for enabling additional temporal layers. Since these are
81 // thresholds in between layers, we have one additional layer. One threshold
82 // gives two temporal layers, one below the threshold and one above, two give
83 // three, and so on.
84 // The VideoEncoder may redistribute bitrates over the temporal layers so a
85 // bitrate threshold of 100k and an estimate of 105k does not imply that we
86 // get 100k in one temporal layer and 5k in the other, just that the bitrate
87 // in the first temporal layer should not exceed 100k.
88 // TODO(pbos): Apart from a special case for two-layer screencast these
89 // thresholds are not propagated to the VideoEncoder. To be implemented.
90 std::vector<int> temporal_layer_thresholds_bps;
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000091};
92
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +000093struct VideoEncoderConfig {
Erik Språng143cec12015-04-28 10:01:41 +020094 enum class ContentType {
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +000095 kRealtimeVideo,
Erik Språng143cec12015-04-28 10:01:41 +020096 kScreen,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +000097 };
98
kwiberg@webrtc.orgac2d27d2015-02-26 13:59:22 +000099 VideoEncoderConfig();
100 ~VideoEncoderConfig();
pbos@webrtc.orgad3b5a52014-10-24 09:23:21 +0000101 std::string ToString() const;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000102
103 std::vector<VideoStream> streams;
104 ContentType content_type;
105 void* encoder_specific_settings;
pbos@webrtc.orgad3b5a52014-10-24 09:23:21 +0000106
107 // Padding will be used up to this bitrate regardless of the bitrate produced
108 // by the encoder. Padding above what's actually produced by the encoder helps
109 // maintaining a higher bitrate estimate. Padding will however not be sent
110 // unless the estimated bandwidth indicates that the link can handle it.
111 int min_transmit_bitrate_bps;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000112};
113
mflodman@webrtc.org65f995a2013-04-18 12:02:52 +0000114} // namespace webrtc
115
pbos@webrtc.org3c107582014-07-20 15:27:35 +0000116#endif // WEBRTC_CONFIG_H_