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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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64 RTC_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 if (first_packet_received_()) { | 73 if (first_packet_received_()) { |
74 LOG(LS_INFO) << "Received first video RTP packet"; | 74 LOG(LS_ERROR) << "Received first video RTP packet"; |
75 } | 75 } |
76 | 76 |
77 // We are not allowed to hold a critical section when calling below functions. | 77 // We are not allowed to hold a critical section when calling below functions. |
78 std::unique_ptr<RtpDepacketizer> depacketizer( | 78 std::unique_ptr<RtpDepacketizer> depacketizer( |
79 RtpDepacketizer::Create(rtp_header->type.Video.codec)); | 79 RtpDepacketizer::Create(rtp_header->type.Video.codec)); |
80 if (depacketizer.get() == NULL) { | 80 if (depacketizer.get() == NULL) { |
81 LOG(LS_ERROR) << "Failed to create depacketizer."; | 81 LOG(LS_ERROR) << "Failed to create depacketizer."; |
82 return -1; | 82 return -1; |
83 } | 83 } |
84 | 84 |
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128 RtpFeedback* callback, | 128 RtpFeedback* callback, |
129 int8_t payload_type, | 129 int8_t payload_type, |
130 const char payload_name[RTP_PAYLOAD_NAME_SIZE], | 130 const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
131 const PayloadUnion& specific_payload) const { | 131 const PayloadUnion& specific_payload) const { |
132 // TODO(pbos): Remove as soon as audio can handle a changing payload type | 132 // TODO(pbos): Remove as soon as audio can handle a changing payload type |
133 // without this callback. | 133 // without this callback. |
134 return 0; | 134 return 0; |
135 } | 135 } |
136 | 136 |
137 } // namespace webrtc | 137 } // namespace webrtc |
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