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

Issue 1335923002: Add RTC_ prefix to (D)CHECKs and related macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 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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54 bool is_red, 54 bool is_red,
55 const uint8_t* payload, 55 const uint8_t* payload,
56 size_t payload_length, 56 size_t payload_length,
57 int64_t timestamp_ms, 57 int64_t timestamp_ms,
58 bool is_first_packet) { 58 bool is_first_packet) {
59 TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp", 59 TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
60 "seqnum", rtp_header->header.sequenceNumber, "timestamp", 60 "seqnum", rtp_header->header.sequenceNumber, "timestamp",
61 rtp_header->header.timestamp); 61 rtp_header->header.timestamp);
62 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType; 62 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
63 63
64 DCHECK_GE(payload_length, rtp_header->header.paddingLength); 64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
65 const size_t payload_data_length = 65 const size_t payload_data_length =
66 payload_length - rtp_header->header.paddingLength; 66 payload_length - rtp_header->header.paddingLength;
67 67
68 if (payload == NULL || payload_data_length == 0) { 68 if (payload == NULL || payload_data_length == 0) {
69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
70 : -1; 70 : -1;
71 } 71 }
72 72
73 // We are not allowed to hold a critical section when calling below functions. 73 // We are not allowed to hold a critical section when calling below functions.
74 rtc::scoped_ptr<RtpDepacketizer> depacketizer( 74 rtc::scoped_ptr<RtpDepacketizer> depacketizer(
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120 callback->OnInitializeDecoder( 120 callback->OnInitializeDecoder(
121 id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) { 121 id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
122 LOG(LS_ERROR) << "Failed to created decoder for payload type: " 122 LOG(LS_ERROR) << "Failed to created decoder for payload type: "
123 << static_cast<int>(payload_type); 123 << static_cast<int>(payload_type);
124 return -1; 124 return -1;
125 } 125 }
126 return 0; 126 return 0;
127 } 127 }
128 128
129 } // namespace webrtc 129 } // namespace webrtc
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