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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 #ifndef MEDIA_CAST_AUDIO_SENDER_H_ | |
6 #define MEDIA_CAST_AUDIO_SENDER_H_ | |
7 | |
8 #include "base/callback.h" | |
9 #include "base/memory/ref_counted.h" | |
10 #include "base/memory/scoped_ptr.h" | |
11 #include "base/memory/weak_ptr.h" | |
12 #include "base/threading/non_thread_safe.h" | |
13 #include "base/time/tick_clock.h" | |
14 #include "base/time/time.h" | |
15 #include "media/base/audio_bus.h" | |
16 #include "media/cast/cast_config.h" | |
17 #include "media/cast/cast_environment.h" | |
18 #include "media/cast/logging/logging_defines.h" | |
19 #include "media/cast/rtcp/rtcp.h" | |
20 #include "media/cast/rtp_timestamp_helper.h" | |
21 | |
22 namespace media { | |
23 namespace cast { | |
24 | |
25 class AudioEncoder; | |
26 | |
27 // Not thread safe. Only called from the main cast thread. | |
28 // This class owns all objects related to sending audio, objects that create RTP | |
29 // packets, congestion control, audio encoder, parsing and sending of | |
30 // RTCP packets. | |
31 // Additionally it posts a bunch of delayed tasks to the main thread for various | |
32 // timeouts. | |
33 class AudioSender : public RtcpSenderFeedback, | |
34 public base::NonThreadSafe, | |
35 public base::SupportsWeakPtr<AudioSender> { | |
36 public: | |
37 AudioSender(scoped_refptr<CastEnvironment> cast_environment, | |
38 const AudioSenderConfig& audio_config, | |
39 transport::CastTransportSender* const transport_sender); | |
40 | |
41 virtual ~AudioSender(); | |
42 | |
43 CastInitializationStatus InitializationResult() const { | |
44 return cast_initialization_status_; | |
45 } | |
46 | |
47 // Note: It is not guaranteed that |audio_frame| will actually be encoded and | |
48 // sent, if AudioSender detects too many frames in flight. Therefore, clients | |
49 // should be careful about the rate at which this method is called. | |
50 // | |
51 // Note: It is invalid to call this method if InitializationResult() returns | |
52 // anything but STATUS_AUDIO_INITIALIZED. | |
53 void InsertAudio(scoped_ptr<AudioBus> audio_bus, | |
54 const base::TimeTicks& recorded_time); | |
55 | |
56 // Only called from the main cast thread. | |
57 void IncomingRtcpPacket(scoped_ptr<Packet> packet); | |
58 | |
59 protected: | |
60 // Protected for testability. | |
61 virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) | |
62 OVERRIDE; | |
63 | |
64 private: | |
65 // Schedule and execute periodic sending of RTCP report. | |
66 void ScheduleNextRtcpReport(); | |
67 void SendRtcpReport(bool schedule_future_reports); | |
68 | |
69 // Schedule and execute periodic checks for re-sending packets. If no | |
70 // acknowledgements have been received for "too long," AudioSender will | |
71 // speculatively re-send certain packets of an unacked frame to kick-start | |
72 // re-transmission. This is a last resort tactic to prevent the session from | |
73 // getting stuck after a long outage. | |
74 void ScheduleNextResendCheck(); | |
75 void ResendCheck(); | |
76 void ResendForKickstart(); | |
77 | |
78 // Returns true if there are too many frames in flight, as defined by the | |
79 // configured target playout delay plus simple logic. When this is true, | |
80 // InsertAudio() will silenty drop frames instead of sending them to the audio | |
81 // encoder. | |
82 bool AreTooManyFramesInFlight() const; | |
83 | |
84 // Called by the |audio_encoder_| with the next EncodedFrame to send. | |
85 void SendEncodedAudioFrame(scoped_ptr<transport::EncodedFrame> audio_frame); | |
86 | |
87 const scoped_refptr<CastEnvironment> cast_environment_; | |
88 | |
89 // The total amount of time between a frame's capture/recording on the sender | |
90 // and its playback on the receiver (i.e., shown to a user). This is fixed as | |
91 // a value large enough to give the system sufficient time to encode, | |
92 // transmit/retransmit, receive, decode, and render; given its run-time | |
93 // environment (sender/receiver hardware performance, network conditions, | |
94 // etc.). | |
95 const base::TimeDelta target_playout_delay_; | |
96 | |
97 // Sends encoded frames over the configured transport (e.g., UDP). In | |
98 // Chromium, this could be a proxy that first sends the frames from a renderer | |
99 // process to the browser process over IPC, with the browser process being | |
100 // responsible for "packetizing" the frames and pushing packets into the | |
101 // network layer. | |
102 transport::CastTransportSender* const transport_sender_; | |
103 | |
104 // Maximum number of outstanding frames before the encoding and sending of | |
105 // new frames shall halt. | |
106 const int max_unacked_frames_; | |
107 | |
108 // Encodes AudioBuses into EncodedFrames. | |
109 scoped_ptr<AudioEncoder> audio_encoder_; | |
110 const int configured_encoder_bitrate_; | |
111 | |
112 // Manages sending/receiving of RTCP packets, including sender/receiver | |
113 // reports. | |
114 Rtcp rtcp_; | |
115 | |
116 // Records lip-sync (i.e., mapping of RTP <--> NTP timestamps), and | |
117 // extrapolates this mapping to any other point in time. | |
118 RtpTimestampHelper rtp_timestamp_helper_; | |
119 | |
120 // Counts how many RTCP reports are being "aggressively" sent (i.e., one per | |
121 // frame) at the start of the session. Once a threshold is reached, RTCP | |
122 // reports are instead sent at the configured interval + random drift. | |
123 int num_aggressive_rtcp_reports_sent_; | |
124 | |
125 // This is "null" until the first frame is sent. Thereafter, this tracks the | |
126 // last time any frame was sent or re-sent. | |
127 base::TimeTicks last_send_time_; | |
128 | |
129 // The ID of the last frame sent. Logic throughout AudioSender assumes this | |
130 // can safely wrap-around. This member is invalid until | |
131 // |!last_send_time_.is_null()|. | |
132 uint32 last_sent_frame_id_; | |
133 | |
134 // The ID of the latest (not necessarily the last) frame that has been | |
135 // acknowledged. Logic throughout AudioSender assumes this can safely | |
136 // wrap-around. This member is invalid until |!last_send_time_.is_null()|. | |
137 uint32 latest_acked_frame_id_; | |
138 | |
139 // Counts the number of duplicate ACK that are being received. When this | |
140 // number reaches a threshold, the sender will take this as a sign that the | |
141 // receiver hasn't yet received the first packet of the next frame. In this | |
142 // case, AudioSender will trigger a re-send of the next frame. | |
143 int duplicate_ack_counter_; | |
144 | |
145 // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED. | |
146 CastInitializationStatus cast_initialization_status_; | |
147 | |
148 // This is a "good enough" mapping for finding the RTP timestamp associated | |
149 // with a video frame. The key is the lowest 8 bits of frame id (which is | |
150 // what is sent via RTCP). This map is used for logging purposes. | |
151 RtpTimestamp frame_id_to_rtp_timestamp_[256]; | |
152 | |
153 // NOTE: Weak pointers must be invalidated before all other member variables. | |
154 base::WeakPtrFactory<AudioSender> weak_factory_; | |
155 | |
156 DISALLOW_COPY_AND_ASSIGN(AudioSender); | |
157 }; | |
158 | |
159 } // namespace cast | |
160 } // namespace media | |
161 | |
162 #endif // MEDIA_CAST_AUDIO_SENDER_H_ | |
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