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Side by Side Diff: media/cast/audio_sender/audio_sender.h

Issue 388663003: Cast: Reshuffle files under media/cast (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: missing includes Created 6 years, 5 months ago
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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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