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Side by Side Diff: media/cast/transport/rtp_sender/rtp_sender.cc

Issue 288103002: [Cast] EncodedAudioFrame+EncodedVideoFrame+reference_time --> EncodedFrame (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 6 years, 7 months ago
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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/transport/rtp_sender/rtp_sender.h" 5 #include "media/cast/transport/rtp_sender/rtp_sender.h"
6 6
7 #include "base/logging.h" 7 #include "base/logging.h"
8 #include "base/rand_util.h" 8 #include "base/rand_util.h"
9 #include "media/cast/transport/cast_transport_defines.h" 9 #include "media/cast/transport/cast_transport_defines.h"
10 #include "media/cast/transport/pacing/paced_sender.h" 10 #include "media/cast/transport/pacing/paced_sender.h"
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54 } 54 }
55 config_.audio = false; 55 config_.audio = false;
56 config_.ssrc = config.rtp.config.ssrc; 56 config_.ssrc = config.rtp.config.ssrc;
57 config_.payload_type = config.rtp.config.payload_type; 57 config_.payload_type = config.rtp.config.payload_type;
58 config_.frequency = kVideoFrequency; 58 config_.frequency = kVideoFrequency;
59 config_.video_codec = config.codec; 59 config_.video_codec = config.codec;
60 packetizer_.reset(new RtpPacketizer(transport_, storage_.get(), config_)); 60 packetizer_.reset(new RtpPacketizer(transport_, storage_.get(), config_));
61 return true; 61 return true;
62 } 62 }
63 63
64 void RtpSender::IncomingEncodedVideoFrame(const EncodedVideoFrame* video_frame, 64 void RtpSender::SendFrame(const EncodedFrame& frame) {
65 const base::TimeTicks& capture_time) {
66 DCHECK(packetizer_); 65 DCHECK(packetizer_);
67 packetizer_->IncomingEncodedVideoFrame(video_frame, capture_time); 66 packetizer_->SendFrameAsPackets(frame);
68 }
69
70 void RtpSender::IncomingEncodedAudioFrame(
71 const EncodedAudioFrame* audio_frame,
72 const base::TimeTicks& recorded_time) {
73 DCHECK(packetizer_);
74 packetizer_->IncomingEncodedAudioFrame(audio_frame, recorded_time);
75 } 67 }
76 68
77 void RtpSender::ResendPackets( 69 void RtpSender::ResendPackets(
78 const MissingFramesAndPacketsMap& missing_frames_and_packets) { 70 const MissingFramesAndPacketsMap& missing_frames_and_packets) {
79 DCHECK(storage_); 71 DCHECK(storage_);
80 // Iterate over all frames in the list. 72 // Iterate over all frames in the list.
81 for (MissingFramesAndPacketsMap::const_iterator it = 73 for (MissingFramesAndPacketsMap::const_iterator it =
82 missing_frames_and_packets.begin(); 74 missing_frames_and_packets.begin();
83 it != missing_frames_and_packets.end(); 75 it != missing_frames_and_packets.end();
84 ++it) { 76 ++it) {
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162 packetizer_->LastSentTimestamp(&time_sent, &rtp_timestamp); 154 packetizer_->LastSentTimestamp(&time_sent, &rtp_timestamp);
163 sender_info.send_packet_count = packetizer_->send_packets_count(); 155 sender_info.send_packet_count = packetizer_->send_packets_count();
164 sender_info.send_octet_count = packetizer_->send_octet_count(); 156 sender_info.send_octet_count = packetizer_->send_octet_count();
165 stats_callback_.Run(sender_info, time_sent, rtp_timestamp); 157 stats_callback_.Run(sender_info, time_sent, rtp_timestamp);
166 ScheduleNextStatsReport(); 158 ScheduleNextStatsReport();
167 } 159 }
168 160
169 } // namespace transport 161 } // namespace transport
170 } // namespace cast 162 } // namespace cast
171 } // namespace media 163 } // namespace media
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