Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(494)

Side by Side Diff: webrtc/modules/include/module_common_types.h

Issue 2772033002: Add content type information to encoded images and corresponding rtp extension header (Closed)
Patch Set: Fix typo, leading to failed video catpure test Created 3 years, 8 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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
(...skipping 40 matching lines...) Expand 10 before | Expand all | Expand 10 after
51 }; 51 };
52 // Since RTPVideoHeader is used as a member of a union, it can't have a 52 // Since RTPVideoHeader is used as a member of a union, it can't have a
53 // non-trivial default constructor. 53 // non-trivial default constructor.
54 struct RTPVideoHeader { 54 struct RTPVideoHeader {
55 uint16_t width; // size 55 uint16_t width; // size
56 uint16_t height; 56 uint16_t height;
57 VideoRotation rotation; 57 VideoRotation rotation;
58 58
59 PlayoutDelay playout_delay; 59 PlayoutDelay playout_delay;
60 60
61 VideoContentType content_type;
62
61 union { 63 union {
62 bool is_first_packet_in_frame; 64 bool is_first_packet_in_frame;
63 RTC_DEPRECATED bool isFirstPacket; // first packet in frame 65 RTC_DEPRECATED bool isFirstPacket; // first packet in frame
64 }; 66 };
65 uint8_t simulcastIdx; // Index if the simulcast encoder creating 67 uint8_t simulcastIdx; // Index if the simulcast encoder creating
66 // this frame, 0 if not using simulcast. 68 // this frame, 0 if not using simulcast.
67 RtpVideoCodecTypes codec; 69 RtpVideoCodecTypes codec;
68 RTPVideoTypeHeader codecHeader; 70 RTPVideoTypeHeader codecHeader;
69 }; 71 };
70 union RTPTypeHeader { 72 union RTPTypeHeader {
71 RTPAudioHeader Audio; 73 RTPAudioHeader Audio;
72 RTPVideoHeader Video; 74 RTPVideoHeader Video;
73 }; 75 };
74 76
75 struct WebRtcRTPHeader { 77 struct WebRtcRTPHeader {
76 RTPHeader header; 78 RTPHeader header;
77 FrameType frameType; 79 FrameType frameType;
78 RTPTypeHeader type; 80 RTPTypeHeader type;
79 // NTP time of the capture time in local timebase in milliseconds. 81 // NTP time of the capture time in local timebase in milliseconds.
80 int64_t ntp_time_ms; 82 int64_t ntp_time_ms;
81 }; 83 };
82 84
83 class RTPFragmentationHeader { 85 class RTPFragmentationHeader {
84 public: 86 public:
85 RTPFragmentationHeader() 87 RTPFragmentationHeader()
86 : fragmentationVectorSize(0), 88 : fragmentationVectorSize(0),
87 fragmentationOffset(NULL), 89 fragmentationOffset(NULL),
88 fragmentationLength(NULL), 90 fragmentationLength(NULL),
89 fragmentationTimeDiff(NULL), 91 fragmentationTimeDiff(NULL),
90 fragmentationPlType(NULL) {}; 92 fragmentationPlType(NULL) {}
91 93
92 ~RTPFragmentationHeader() { 94 ~RTPFragmentationHeader() {
93 delete[] fragmentationOffset; 95 delete[] fragmentationOffset;
94 delete[] fragmentationLength; 96 delete[] fragmentationLength;
95 delete[] fragmentationTimeDiff; 97 delete[] fragmentationTimeDiff;
96 delete[] fragmentationPlType; 98 delete[] fragmentationPlType;
97 } 99 }
98 100
99 void CopyFrom(const RTPFragmentationHeader& src) { 101 void CopyFrom(const RTPFragmentationHeader& src) {
100 if (this == &src) { 102 if (this == &src) {
(...skipping 467 matching lines...) Expand 10 before | Expand all | Expand 10 after
568 static constexpr int kNotAProbe = -1; 570 static constexpr int kNotAProbe = -1;
569 int send_bitrate_bps = -1; 571 int send_bitrate_bps = -1;
570 int probe_cluster_id = kNotAProbe; 572 int probe_cluster_id = kNotAProbe;
571 int probe_cluster_min_probes = -1; 573 int probe_cluster_min_probes = -1;
572 int probe_cluster_min_bytes = -1; 574 int probe_cluster_min_bytes = -1;
573 }; 575 };
574 576
575 } // namespace webrtc 577 } // namespace webrtc
576 578
577 #endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ 579 #endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
OLDNEW
« no previous file with comments | « webrtc/media/engine/webrtcvideoengine2_unittest.cc ('k') | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698