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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_packetizer/rtp_packetizer.h" | |
6 | |
7 #include "base/logging.h" | |
8 #include "media/cast/cast_defines.h" | |
9 #include "media/cast/pacing/paced_sender.h" | |
10 #include "net/base/big_endian.h" | |
11 | |
12 namespace media { | |
13 namespace cast { | |
14 | |
15 static const uint16 kCommonRtpHeaderLength = 12; | |
16 static const uint16 kCastRtpHeaderLength = 7; | |
17 static const uint8 kCastKeyFrameBitMask = 0x80; | |
18 static const uint8 kCastReferenceFrameIdBitMask = 0x40; | |
19 | |
20 RtpPacketizer::RtpPacketizer(PacedPacketSender* transport, | |
21 PacketStorage* packet_storage, | |
22 RtpPacketizerConfig rtp_packetizer_config) | |
23 : config_(rtp_packetizer_config), | |
24 transport_(transport), | |
25 packet_storage_(packet_storage), | |
26 sequence_number_(config_.sequence_number), | |
27 rtp_timestamp_(config_.rtp_timestamp), | |
28 packet_id_(0), | |
29 send_packets_count_(0), | |
30 send_octet_count_(0) { | |
31 DCHECK(transport) << "Invalid argument"; | |
32 } | |
33 | |
34 RtpPacketizer::~RtpPacketizer() {} | |
35 | |
36 void RtpPacketizer::IncomingEncodedVideoFrame( | |
37 const EncodedVideoFrame* video_frame, | |
38 const base::TimeTicks& capture_time) { | |
39 DCHECK(!config_.audio) << "Invalid state"; | |
40 if (config_.audio) return; | |
41 | |
42 // Timestamp is in 90 KHz for video. | |
43 rtp_timestamp_ = GetVideoRtpTimestamp(capture_time); | |
44 time_last_sent_rtp_timestamp_ = capture_time; | |
45 | |
46 Cast(video_frame->key_frame, | |
47 video_frame->frame_id, | |
48 video_frame->last_referenced_frame_id, | |
49 rtp_timestamp_, | |
50 video_frame->data); | |
51 } | |
52 | |
53 void RtpPacketizer::IncomingEncodedAudioFrame( | |
54 const EncodedAudioFrame* audio_frame, | |
55 const base::TimeTicks& recorded_time) { | |
56 DCHECK(config_.audio) << "Invalid state"; | |
57 if (!config_.audio) return; | |
58 | |
59 rtp_timestamp_ += audio_frame->samples; // Timestamp is in samples for audio. | |
60 time_last_sent_rtp_timestamp_ = recorded_time; | |
61 Cast(true, audio_frame->frame_id, 0, rtp_timestamp_, audio_frame->data); | |
62 } | |
63 | |
64 uint16 RtpPacketizer::NextSequenceNumber() { | |
65 ++sequence_number_; | |
66 return sequence_number_ - 1; | |
67 } | |
68 | |
69 bool RtpPacketizer::LastSentTimestamp(base::TimeTicks* time_sent, | |
70 uint32* rtp_timestamp) const { | |
71 if (time_last_sent_rtp_timestamp_.is_null()) return false; | |
72 | |
73 *time_sent = time_last_sent_rtp_timestamp_; | |
74 *rtp_timestamp = rtp_timestamp_; | |
75 return true; | |
76 } | |
77 | |
78 // TODO(mikhal): Switch to pass data with a const_ref. | |
79 void RtpPacketizer::Cast(bool is_key, | |
80 uint32 frame_id, | |
81 uint32 reference_frame_id, | |
82 uint32 timestamp, | |
83 const std::string& data) { | |
84 uint16 rtp_header_length = kCommonRtpHeaderLength + kCastRtpHeaderLength; | |
85 uint16 max_length = config_.max_payload_length - rtp_header_length - 1; | |
86 | |
87 // Split the payload evenly (round number up). | |
88 size_t num_packets = (data.size() + max_length) / max_length; | |
89 size_t payload_length = (data.size() + num_packets) / num_packets; | |
90 DCHECK_LE(payload_length, max_length) << "Invalid argument"; | |
91 | |
92 PacketList packets; | |
93 | |
94 size_t remaining_size = data.size(); | |
95 std::string::const_iterator data_iter = data.begin(); | |
96 while (remaining_size > 0) { | |
97 Packet packet; | |
98 | |
99 if (remaining_size < payload_length) { | |
100 payload_length = remaining_size; | |
101 } | |
102 remaining_size -= payload_length; | |
103 BuildCommonRTPheader(&packet, remaining_size == 0, timestamp); | |
104 | |
105 // Build Cast header. | |
106 packet.push_back( | |
107 (is_key ? kCastKeyFrameBitMask : 0) | kCastReferenceFrameIdBitMask); | |
108 packet.push_back(frame_id); | |
109 size_t start_size = packet.size(); | |
110 packet.resize(start_size + 4); | |
111 net::BigEndianWriter big_endian_writer(&(packet[start_size]), 4); | |
112 big_endian_writer.WriteU16(packet_id_); | |
113 big_endian_writer.WriteU16(static_cast<uint16>(num_packets - 1)); | |
114 packet.push_back(static_cast<uint8>(reference_frame_id)); | |
115 | |
116 // Copy payload data. | |
117 packet.insert(packet.end(), data_iter, data_iter + payload_length); | |
118 | |
119 // Store packet. | |
120 packet_storage_->StorePacket(frame_id, packet_id_, &packet); | |
121 ++packet_id_; | |
122 data_iter += payload_length; | |
123 | |
124 // Update stats. | |
125 ++send_packets_count_; | |
126 send_octet_count_ += payload_length; | |
127 packets.push_back(packet); | |
128 } | |
129 DCHECK(packet_id_ == num_packets) << "Invalid state"; | |
130 | |
131 // Send to network. | |
132 transport_->SendPackets(packets); | |
133 | |
134 // Prepare for next frame. | |
135 packet_id_ = 0; | |
136 } | |
137 | |
138 void RtpPacketizer::BuildCommonRTPheader( | |
139 Packet* packet, bool marker_bit, uint32 time_stamp) { | |
140 packet->push_back(0x80); | |
141 packet->push_back(static_cast<uint8>(config_.payload_type) | | |
142 (marker_bit ? kRtpMarkerBitMask : 0)); | |
143 size_t start_size = packet->size(); | |
144 packet->resize(start_size + 10); | |
145 net::BigEndianWriter big_endian_writer(&((*packet)[start_size]), 10); | |
146 big_endian_writer.WriteU16(sequence_number_); | |
147 big_endian_writer.WriteU32(time_stamp); | |
148 big_endian_writer.WriteU32(config_.ssrc); | |
149 ++sequence_number_; | |
150 } | |
151 | |
152 } // namespace cast | |
153 } // namespace media | |
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