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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc

Issue 2007743003: Add sender controlled playout delay limits (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@cleanup_rtp_hdr_extensions
Patch Set: Created 4 years, 7 months ago
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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
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91 rtp_header->frameType = parsed_payload.frame_type; 91 rtp_header->frameType = parsed_payload.frame_type;
92 rtp_header->type = parsed_payload.type; 92 rtp_header->type = parsed_payload.type;
93 rtp_header->type.Video.rotation = kVideoRotation_0; 93 rtp_header->type.Video.rotation = kVideoRotation_0;
94 94
95 // Retrieve the video rotation information. 95 // Retrieve the video rotation information.
96 if (rtp_header->header.extension.hasVideoRotation) { 96 if (rtp_header->header.extension.hasVideoRotation) {
97 rtp_header->type.Video.rotation = ConvertCVOByteToVideoRotation( 97 rtp_header->type.Video.rotation = ConvertCVOByteToVideoRotation(
98 rtp_header->header.extension.videoRotation); 98 rtp_header->header.extension.videoRotation);
99 } 99 }
100 100
101 rtp_header->type.Video.min_playout_delay_ms =
102 rtp_header->header.extension.min_playout_delay_ms;
103 rtp_header->type.Video.max_playout_delay_ms =
104 rtp_header->header.extension.max_playout_delay_ms;
105
101 return data_callback_->OnReceivedPayloadData(parsed_payload.payload, 106 return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
102 parsed_payload.payload_length, 107 parsed_payload.payload_length,
103 rtp_header) == 0 108 rtp_header) == 0
104 ? 0 109 ? 0
105 : -1; 110 : -1;
106 } 111 }
107 112
108 int RTPReceiverVideo::GetPayloadTypeFrequency() const { 113 int RTPReceiverVideo::GetPayloadTypeFrequency() const {
109 return kVideoPayloadTypeFrequency; 114 return kVideoPayloadTypeFrequency;
110 } 115 }
111 116
112 RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive( 117 RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
113 uint16_t last_payload_length) const { 118 uint16_t last_payload_length) const {
114 return kRtpDead; 119 return kRtpDead;
115 } 120 }
116 121
117 int32_t RTPReceiverVideo::InvokeOnInitializeDecoder( 122 int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
118 RtpFeedback* callback, 123 RtpFeedback* callback,
119 int8_t payload_type, 124 int8_t payload_type,
120 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 125 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
121 const PayloadUnion& specific_payload) const { 126 const PayloadUnion& specific_payload) const {
122 // TODO(pbos): Remove as soon as audio can handle a changing payload type 127 // TODO(pbos): Remove as soon as audio can handle a changing payload type
123 // without this callback. 128 // without this callback.
124 return 0; 129 return 0;
125 } 130 }
126 131
127 } // namespace webrtc 132 } // namespace webrtc
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