1 // Copyright 2014 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.
5 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
6 #define MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
11 #include "base/basictypes.h"
12 #include "base/callback.h"
13 #include "base/memory/linked_ptr.h"
14 #include "base/memory/ref_counted.h"
15 #include "base/stl_util.h"
16 #include "media/cast/net/cast_transport_defines.h"
28 CODEC_LAST = CODEC_VIDEO_H264
31 struct CastTransportRtpConfig {
32 CastTransportRtpConfig();
33 ~CastTransportRtpConfig();
35 // Identifier refering to this sender.
38 // Identifier for incoming RTCP traffic.
41 // RTP payload type enum: Specifies the type/encoding of frame data.
44 // The number of most-recent frames that must be stored in the transport
45 // layer, to facilitate re-transmissions.
48 // The AES crypto key and initialization vector. Each of these strings
49 // contains the data in binary form, of size kAesKeySize. If they are empty
50 // strings, crypto is not being used.
52 std::string aes_iv_mask;
55 // A combination of metadata and data for one encoded frame. This can contain
56 // audio data or video data or other.
59 // "null" value, used to indicate whether |dependency| has been set.
62 // Not decodable without the reference frame indicated by
63 // |referenced_frame_id|.
66 // Independently decodable.
69 // Independently decodable, and no future frames will depend on any frames
79 // Convenience accessors to data as an array of uint8 elements.
80 const uint8* bytes() const {
81 return reinterpret_cast<uint8*>(string_as_array(
82 const_cast<std::string*>(&data)));
84 uint8* mutable_bytes() {
85 return reinterpret_cast<uint8*>(string_as_array(&data));
88 // Copies all data members except |data| to |dest|.
89 // Does not modify |dest->data|.
90 void CopyMetadataTo(EncodedFrame* dest) const;
92 // This frame's dependency relationship with respect to other frames.
93 Dependency dependency;
95 // The label associated with this frame. Implies an ordering relative to
96 // other frames in the same stream.
99 // The label associated with the frame upon which this frame depends. If
100 // this frame does not require any other frame in order to become decodable
101 // (e.g., key frames), |referenced_frame_id| must equal |frame_id|.
102 uint32 referenced_frame_id;
104 // The stream timestamp, on the timeline of the signal data. For example, RTP
105 // timestamps for audio are usually defined as the total number of audio
106 // samples encoded in all prior frames. A playback system uses this value to
107 // detect gaps in the stream, and otherwise stretch the signal to match
109 uint32 rtp_timestamp;
111 // The common reference clock timestamp for this frame. This value originates
112 // from a sender and is used to provide lip synchronization between streams in
113 // a receiver. Thus, in the sender context, this is set to the time at which
114 // the frame was captured/recorded. In the receiver context, this is set to
115 // the target playout time. Over a sequence of frames, this time value is
116 // expected to drift with respect to the elapsed time implied by the RTP
117 // timestamps; and it may not necessarily increment with precise regularity.
118 base::TimeTicks reference_time;
120 // The encoded signal data.
124 typedef std::vector<uint8> Packet;
125 typedef scoped_refptr<base::RefCountedData<Packet> > PacketRef;
126 typedef std::vector<PacketRef> PacketList;
128 typedef base::Callback<void(scoped_ptr<Packet> packet)> PacketReceiverCallback;
132 // Send a packet to the network. Returns false if the network is blocked
133 // and we should wait for |cb| to be called. It is not allowed to called
134 // SendPacket again until |cb| has been called. Any other errors that
135 // occur will be reported through side channels, in such cases, this function
136 // will return true indicating that the channel is not blocked.
137 virtual bool SendPacket(PacketRef packet, const base::Closure& cb) = 0;
138 virtual ~PacketSender() {}
141 struct RtcpSenderInfo {
144 // First three members are used for lipsync.
145 // First two members are used for rtt.
148 uint32 rtp_timestamp;
149 uint32 send_packet_count;
150 size_t send_octet_count;
153 struct RtcpReportBlock {
156 uint32 remote_ssrc; // SSRC of sender of this report.
157 uint32 media_ssrc; // SSRC of the RTP packet sender.
159 uint32 cumulative_lost; // 24 bits valid.
160 uint32 extended_high_sequence_number;
163 uint32 delay_since_last_sr;
166 struct RtcpDlrrReportBlock {
167 RtcpDlrrReportBlock();
168 ~RtcpDlrrReportBlock();
170 uint32 delay_since_last_rr;
173 inline bool operator==(RtcpSenderInfo lhs, RtcpSenderInfo rhs) {
174 return lhs.ntp_seconds == rhs.ntp_seconds &&
175 lhs.ntp_fraction == rhs.ntp_fraction &&
176 lhs.rtp_timestamp == rhs.rtp_timestamp &&
177 lhs.send_packet_count == rhs.send_packet_count &&
178 lhs.send_octet_count == rhs.send_octet_count;
184 #endif // MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_