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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "media/cast/audio_receiver/audio_receiver.h" | 5 #include "media/cast/audio_receiver/audio_receiver.h" |
6 | 6 |
7 #include <algorithm> | 7 #include <algorithm> |
8 | 8 |
9 #include "base/bind.h" | 9 #include "base/bind.h" |
10 #include "base/logging.h" | 10 #include "base/logging.h" |
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70 | 70 |
71 void AudioReceiver::OnReceivedPayloadData(const uint8* payload_data, | 71 void AudioReceiver::OnReceivedPayloadData(const uint8* payload_data, |
72 size_t payload_size, | 72 size_t payload_size, |
73 const RtpCastHeader& rtp_header) { | 73 const RtpCastHeader& rtp_header) { |
74 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 74 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
75 base::TimeTicks now = cast_environment_->Clock()->NowTicks(); | 75 base::TimeTicks now = cast_environment_->Clock()->NowTicks(); |
76 | 76 |
77 // TODO(pwestin): update this as video to refresh over time. | 77 // TODO(pwestin): update this as video to refresh over time. |
78 if (time_first_incoming_packet_.is_null()) { | 78 if (time_first_incoming_packet_.is_null()) { |
79 InitializeTimers(); | 79 InitializeTimers(); |
80 first_incoming_rtp_timestamp_ = rtp_header.webrtc.header.timestamp; | 80 first_incoming_rtp_timestamp_ = rtp_header.rtp_timestamp; |
81 time_first_incoming_packet_ = now; | 81 time_first_incoming_packet_ = now; |
82 } | 82 } |
83 | 83 |
84 frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] = | 84 frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] = |
85 rtp_header.webrtc.header.timestamp; | 85 rtp_header.rtp_timestamp; |
86 cast_environment_->Logging()->InsertPacketEvent( | 86 cast_environment_->Logging()->InsertPacketEvent( |
87 now, kAudioPacketReceived, rtp_header.webrtc.header.timestamp, | 87 now, kAudioPacketReceived, rtp_header.rtp_timestamp, |
88 rtp_header.frame_id, rtp_header.packet_id, rtp_header.max_packet_id, | 88 rtp_header.frame_id, rtp_header.packet_id, rtp_header.max_packet_id, |
89 payload_size); | 89 payload_size); |
90 | 90 |
91 bool duplicate = false; | 91 bool duplicate = false; |
92 const bool complete = | 92 const bool complete = |
93 framer_.InsertPacket(payload_data, payload_size, rtp_header, &duplicate); | 93 framer_.InsertPacket(payload_data, payload_size, rtp_header, &duplicate); |
94 if (duplicate) { | 94 if (duplicate) { |
95 cast_environment_->Logging()->InsertPacketEvent( | 95 cast_environment_->Logging()->InsertPacketEvent( |
96 now, | 96 now, |
97 kDuplicateAudioPacketReceived, | 97 kDuplicateAudioPacketReceived, |
98 rtp_header.webrtc.header.timestamp, | 98 rtp_header.rtp_timestamp, |
99 rtp_header.frame_id, | 99 rtp_header.frame_id, |
100 rtp_header.packet_id, | 100 rtp_header.packet_id, |
101 rtp_header.max_packet_id, | 101 rtp_header.max_packet_id, |
102 payload_size); | 102 payload_size); |
103 // Duplicate packets are ignored. | 103 // Duplicate packets are ignored. |
104 return; | 104 return; |
105 } | 105 } |
106 if (!complete) | 106 if (!complete) |
107 return; | 107 return; |
108 | 108 |
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373 } | 373 } |
374 | 374 |
375 void AudioReceiver::SendNextCastMessage() { | 375 void AudioReceiver::SendNextCastMessage() { |
376 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 376 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
377 framer_.SendCastMessage(); // Will only send a message if it is time. | 377 framer_.SendCastMessage(); // Will only send a message if it is time. |
378 ScheduleNextCastMessage(); | 378 ScheduleNextCastMessage(); |
379 } | 379 } |
380 | 380 |
381 } // namespace cast | 381 } // namespace cast |
382 } // namespace media | 382 } // namespace media |
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