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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 | |
3 // found in the LICENSE file. | |
4 | |
5 #include "media/cast/rtp_sender/rtp_sender.h" | |
6 | |
7 #include "base/logging.h" | |
8 #include "base/rand_util.h" | |
9 #include "media/cast/cast_defines.h" | |
10 #include "media/cast/pacing/paced_sender.h" | |
11 #include "media/cast/rtcp/rtcp_defines.h" | |
12 #include "net/base/big_endian.h" | |
13 | |
14 namespace media { | |
15 namespace cast { | |
16 | |
17 RtpSender::RtpSender(scoped_refptr<CastEnvironment> cast_environment, | |
18 const AudioSenderConfig* audio_config, | |
19 const VideoSenderConfig* video_config, | |
20 PacedPacketSender* transport) | |
21 : cast_environment_(cast_environment), | |
22 config_(), | |
23 transport_(transport) { | |
24 // Store generic cast config and create packetizer config. | |
25 DCHECK(audio_config || video_config) << "Invalid argument"; | |
26 if (audio_config) { | |
27 storage_.reset(new PacketStorage(cast_environment->Clock(), | |
28 audio_config->rtp_history_ms)); | |
29 config_.audio = true; | |
30 config_.ssrc = audio_config->sender_ssrc; | |
31 config_.payload_type = audio_config->rtp_payload_type; | |
32 config_.frequency = audio_config->frequency; | |
33 config_.audio_codec = audio_config->codec; | |
34 } else { | |
35 storage_.reset(new PacketStorage(cast_environment->Clock(), | |
36 video_config->rtp_history_ms)); | |
37 config_.audio = false; | |
38 config_.ssrc = video_config->sender_ssrc; | |
39 config_.payload_type = video_config->rtp_payload_type; | |
40 config_.frequency = kVideoFrequency; | |
41 config_.video_codec = video_config->codec; | |
42 } | |
43 // Randomly set start values. | |
44 config_.sequence_number = base::RandInt(0, 65535); | |
45 config_.rtp_timestamp = base::RandInt(0, 65535); | |
46 config_.rtp_timestamp += base::RandInt(0, 65535) << 16; | |
47 packetizer_.reset(new RtpPacketizer(transport, storage_.get(), config_)); | |
48 } | |
49 | |
50 RtpSender::~RtpSender() {} | |
51 | |
52 void RtpSender::IncomingEncodedVideoFrame(const EncodedVideoFrame* video_frame, | |
53 const base::TimeTicks& capture_time) { | |
54 packetizer_->IncomingEncodedVideoFrame(video_frame, capture_time); | |
55 } | |
56 | |
57 void RtpSender::IncomingEncodedAudioFrame(const EncodedAudioFrame* audio_frame, | |
58 const base::TimeTicks& recorded_time) { | |
59 packetizer_->IncomingEncodedAudioFrame(audio_frame, recorded_time); | |
60 } | |
61 | |
62 void RtpSender::ResendPackets( | |
63 const MissingFramesAndPacketsMap& missing_frames_and_packets) { | |
64 // Iterate over all frames in the list. | |
65 for (MissingFramesAndPacketsMap::const_iterator it = | |
66 missing_frames_and_packets.begin(); | |
67 it != missing_frames_and_packets.end(); ++it) { | |
68 PacketList packets_to_resend; | |
69 uint8 frame_id = it->first; | |
70 const PacketIdSet& packets_set = it->second; | |
71 bool success = false; | |
72 | |
73 if (packets_set.empty()) { | |
74 VLOG(1) << "Missing all packets in frame " << static_cast<int>(frame_id); | |
75 | |
76 uint16 packet_id = 0; | |
77 do { | |
78 // Get packet from storage. | |
79 success = storage_->GetPacket(frame_id, packet_id, &packets_to_resend); | |
80 | |
81 // Resend packet to the network. | |
82 if (success) { | |
83 VLOG(1) << "Resend " << static_cast<int>(frame_id) | |
84 << ":" << packet_id; | |
85 // Set a unique incremental sequence number for every packet. | |
86 Packet& packet = packets_to_resend.back(); | |
87 UpdateSequenceNumber(&packet); | |
88 // Set the size as correspond to each frame. | |
89 ++packet_id; | |
90 } | |
91 } while (success); | |
92 } else { | |
93 // Iterate over all of the packets in the frame. | |
94 for (PacketIdSet::const_iterator set_it = packets_set.begin(); | |
95 set_it != packets_set.end(); ++set_it) { | |
96 uint16 packet_id = *set_it; | |
97 success = storage_->GetPacket(frame_id, packet_id, &packets_to_resend); | |
98 | |
99 // Resend packet to the network. | |
100 if (success) { | |
101 VLOG(1) << "Resend " << static_cast<int>(frame_id) | |
102 << ":" << packet_id; | |
103 Packet& packet = packets_to_resend.back(); | |
104 UpdateSequenceNumber(&packet); | |
105 } | |
106 } | |
107 } | |
108 transport_->ResendPackets(packets_to_resend); | |
109 } | |
110 } | |
111 | |
112 void RtpSender::UpdateSequenceNumber(Packet* packet) { | |
113 uint16 new_sequence_number = packetizer_->NextSequenceNumber(); | |
114 int index = 2; | |
115 (*packet)[index] = (static_cast<uint8>(new_sequence_number)); | |
116 (*packet)[index + 1] =(static_cast<uint8>(new_sequence_number >> 8)); | |
117 } | |
118 | |
119 void RtpSender::RtpStatistics(const base::TimeTicks& now, | |
120 RtcpSenderInfo* sender_info) { | |
121 // The timestamp of this Rtcp packet should be estimated as the timestamp of | |
122 // the frame being captured at this moment. We are calculating that | |
123 // timestamp as the last frame's timestamp + the time since the last frame | |
124 // was captured. | |
125 uint32 ntp_seconds = 0; | |
126 uint32 ntp_fraction = 0; | |
127 ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction); | |
128 sender_info->ntp_seconds = ntp_seconds; | |
129 sender_info->ntp_fraction = ntp_fraction; | |
130 | |
131 base::TimeTicks time_sent; | |
132 uint32 rtp_timestamp; | |
133 if (packetizer_->LastSentTimestamp(&time_sent, &rtp_timestamp)) { | |
134 base::TimeDelta time_since_last_send = now - time_sent; | |
135 sender_info->rtp_timestamp = rtp_timestamp + | |
136 time_since_last_send.InMilliseconds() * (config_.frequency / 1000); | |
137 } else { | |
138 sender_info->rtp_timestamp = 0; | |
139 } | |
140 sender_info->send_packet_count = packetizer_->send_packets_count(); | |
141 sender_info->send_octet_count = packetizer_->send_octet_count(); | |
142 } | |
143 | |
144 } // namespace cast | |
145 } // namespace media | |
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