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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_ | 5 #ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_ |
6 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_ | 6 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_ |
7 | 7 |
8 #include "base/basictypes.h" | 8 #include "base/basictypes.h" |
9 #include "base/callback.h" | 9 #include "base/callback.h" |
10 #include "base/macros.h" | 10 #include "base/macros.h" |
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32 // AudioReceiver receives packets out-of-order while clients make requests for | 32 // AudioReceiver receives packets out-of-order while clients make requests for |
33 // complete frames in-order. (A frame consists of one or more packets.) | 33 // complete frames in-order. (A frame consists of one or more packets.) |
34 // | 34 // |
35 // AudioReceiver also includes logic for computing the playout time for each | 35 // AudioReceiver also includes logic for computing the playout time for each |
36 // frame, accounting for a constant targeted playout delay. The purpose of the | 36 // frame, accounting for a constant targeted playout delay. The purpose of the |
37 // playout delay is to provide a fixed window of time between the capture event | 37 // playout delay is to provide a fixed window of time between the capture event |
38 // on the sender and the playout on the receiver. This is important because | 38 // on the sender and the playout on the receiver. This is important because |
39 // each step of the pipeline (i.e., encode frame, then transmit/retransmit from | 39 // each step of the pipeline (i.e., encode frame, then transmit/retransmit from |
40 // the sender, then receive and re-order packets on the receiver, then decode | 40 // the sender, then receive and re-order packets on the receiver, then decode |
41 // frame) can vary in duration and is typically very hard to predict. | 41 // frame) can vary in duration and is typically very hard to predict. |
42 // Heuristics will determine when the targeted playout delay is insufficient in | |
43 // the current environment; and the receiver can then increase the playout | |
44 // delay, notifying the sender, to account for the extra variance. | |
45 // TODO(miu): Make the last sentence true. http://crbug.com/360111 | |
46 // | 42 // |
47 // Two types of frames can be requested: 1) A frame of decoded audio data; or 2) | 43 // Two types of frames can be requested: 1) A frame of decoded audio data; or 2) |
48 // a frame of still-encoded audio data, to be passed into an external audio | 44 // a frame of still-encoded audio data, to be passed into an external audio |
49 // decoder. Each request for a frame includes a callback which AudioReceiver | 45 // decoder. Each request for a frame includes a callback which AudioReceiver |
50 // guarantees will be called at some point in the future unless the | 46 // guarantees will be called at some point in the future unless the |
51 // AudioReceiver is destroyed. Clients should generally limit the number of | 47 // AudioReceiver is destroyed. Clients should generally limit the number of |
52 // outstanding requests (perhaps to just one or two). | 48 // outstanding requests (perhaps to just one or two). |
53 // | 49 // |
54 // This class is not thread safe. Should only be called from the Main cast | 50 // This class is not thread safe. Should only be called from the Main cast |
55 // thread. | 51 // thread. |
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74 | 70 |
75 // Request an encoded audio frame. | 71 // Request an encoded audio frame. |
76 // | 72 // |
77 // The given |callback| is guaranteed to be run at some point in the future, | 73 // The given |callback| is guaranteed to be run at some point in the future, |
78 // even if to respond with NULL at shutdown time. | 74 // even if to respond with NULL at shutdown time. |
79 void GetEncodedAudioFrame(const AudioFrameEncodedCallback& callback); | 75 void GetEncodedAudioFrame(const AudioFrameEncodedCallback& callback); |
80 | 76 |
81 // Deliver another packet, possibly a duplicate, and possibly out-of-order. | 77 // Deliver another packet, possibly a duplicate, and possibly out-of-order. |
82 void IncomingPacket(scoped_ptr<Packet> packet); | 78 void IncomingPacket(scoped_ptr<Packet> packet); |
83 | 79 |
84 // Update target audio delay used to compute the playout time. Rtcp | |
85 // will also be updated (will be included in all outgoing reports). | |
86 void SetTargetDelay(base::TimeDelta target_delay); | |
87 | |
88 protected: | 80 protected: |
89 friend class AudioReceiverTest; // Invokes OnReceivedPayloadData(). | 81 friend class AudioReceiverTest; // Invokes OnReceivedPayloadData(). |
90 | 82 |
91 virtual void OnReceivedPayloadData(const uint8* payload_data, | 83 virtual void OnReceivedPayloadData(const uint8* payload_data, |
92 size_t payload_size, | 84 size_t payload_size, |
93 const RtpCastHeader& rtp_header) OVERRIDE; | 85 const RtpCastHeader& rtp_header) OVERRIDE; |
94 | 86 |
95 // RtpPayloadFeedback implementation. | 87 // RtpPayloadFeedback implementation. |
96 virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE; | 88 virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE; |
97 | 89 |
98 private: | 90 private: |
99 // Processes ready-to-consume packets from |framer_|, decrypting each packet's | 91 // Processes ready-to-consume packets from |framer_|, decrypting each packet's |
100 // payload data, and then running the enqueued callbacks in order (one for | 92 // payload data, and then running the enqueued callbacks in order (one for |
101 // each packet). This method may post a delayed task to re-invoke itself in | 93 // each packet). This method may post a delayed task to re-invoke itself in |
102 // the future to wait for missing/incomplete frames. | 94 // the future to wait for missing/incomplete frames. |
103 void EmitAvailableEncodedFrames(); | 95 void EmitAvailableEncodedFrames(); |
104 | 96 |
97 // Helper used by EmitAvailableEncodedFrames() to schedule itself to be called | |
98 // again after |wait_time| has elapsed. | |
99 void RetryEmitAfterWaiting(base::TimeDelta wait_time); | |
100 | |
105 // Clears the |is_waiting_for_consecutive_frame_| flag and invokes | 101 // Clears the |is_waiting_for_consecutive_frame_| flag and invokes |
106 // EmitAvailableEncodedFrames(). | 102 // EmitAvailableEncodedFrames(). |
107 void EmitAvailableEncodedFramesAfterWaiting(); | 103 void EmitAvailableEncodedFramesAfterWaiting(); |
108 | 104 |
109 // Feeds an EncodedAudioFrame into |audio_decoder_|. GetRawAudioFrame() uses | 105 // Feeds an EncodedAudioFrame into |audio_decoder_|. GetRawAudioFrame() uses |
110 // this as a callback for GetEncodedAudioFrame(). | 106 // this as a callback for GetEncodedAudioFrame(). |
111 void DecodeEncodedAudioFrame( | 107 void DecodeEncodedAudioFrame( |
112 const AudioFrameDecodedCallback& callback, | 108 const AudioFrameDecodedCallback& callback, |
113 scoped_ptr<transport::EncodedAudioFrame> encoded_frame, | 109 scoped_ptr<transport::EncodedAudioFrame> encoded_frame, |
114 const base::TimeTicks& playout_time); | 110 const base::TimeTicks& playout_time); |
115 | 111 |
116 // Return the playout time based on the current time and rtp timestamp. | 112 // Computes the playout time for a frame with the given |rtp_timestamp|. If |
117 base::TimeTicks GetPlayoutTime(base::TimeTicks now, uint32 rtp_timestamp); | 113 // lip-sync info is not available, a best-guess is returned (a hack). |
118 | 114 base::TimeTicks GetPlayoutTime(uint32 rtp_timestamp) const; |
119 void InitializeTimers(); | |
120 | 115 |
121 // Schedule the next RTCP report. | 116 // Schedule the next RTCP report. |
122 void ScheduleNextRtcpReport(); | 117 void ScheduleNextRtcpReport(); |
123 | 118 |
124 // Actually send the next RTCP report. | 119 // Actually send the next RTCP report. |
125 void SendNextRtcpReport(); | 120 void SendNextRtcpReport(); |
126 | 121 |
127 // Schedule timing for the next cast message. | 122 // Schedule timing for the next cast message. |
128 void ScheduleNextCastMessage(); | 123 void ScheduleNextCastMessage(); |
129 | 124 |
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145 bool is_continuous); | 140 bool is_continuous); |
146 | 141 |
147 const scoped_refptr<CastEnvironment> cast_environment_; | 142 const scoped_refptr<CastEnvironment> cast_environment_; |
148 | 143 |
149 // Subscribes to raw events. | 144 // Subscribes to raw events. |
150 // Processes raw audio events to be sent over to the cast sender via RTCP. | 145 // Processes raw audio events to be sent over to the cast sender via RTCP. |
151 ReceiverRtcpEventSubscriber event_subscriber_; | 146 ReceiverRtcpEventSubscriber event_subscriber_; |
152 | 147 |
153 const transport::AudioCodec codec_; | 148 const transport::AudioCodec codec_; |
154 const int frequency_; | 149 const int frequency_; |
155 base::TimeDelta target_delay_delta_; | 150 base::TimeDelta target_playout_delay_; |
151 bool reports_are_scheduled_; | |
hubbe
2014/05/14 23:12:23
Maybe add comments for these variables?
miu
2014/05/16 22:45:47
Done.
| |
156 Framer framer_; | 152 Framer framer_; |
157 scoped_ptr<AudioDecoder> audio_decoder_; | 153 scoped_ptr<AudioDecoder> audio_decoder_; |
158 Rtcp rtcp_; | 154 Rtcp rtcp_; |
159 base::TimeDelta time_offset_; | |
160 base::TimeTicks time_first_incoming_packet_; | |
161 uint32 first_incoming_rtp_timestamp_; | |
162 transport::TransportEncryptionHandler decryptor_; | 155 transport::TransportEncryptionHandler decryptor_; |
163 | 156 |
164 // Outstanding callbacks to run to deliver on client requests for frames. | 157 // Outstanding callbacks to run to deliver on client requests for frames. |
165 std::list<AudioFrameEncodedCallback> frame_request_queue_; | 158 std::list<AudioFrameEncodedCallback> frame_request_queue_; |
166 | 159 |
167 // True while there's an outstanding task to re-invoke | 160 // True while there's an outstanding task to re-invoke |
168 // EmitAvailableEncodedFrames(). | 161 // EmitAvailableEncodedFrames(). |
169 bool is_waiting_for_consecutive_frame_; | 162 bool is_waiting_to_emit_frames_; |
170 | 163 |
171 // This mapping allows us to log AUDIO_ACK_SENT as a frame event. In addition | 164 // This mapping allows us to log AUDIO_ACK_SENT as a frame event. In addition |
172 // it allows the event to be transmitted via RTCP. | 165 // it allows the event to be transmitted via RTCP. |
173 RtpTimestamp frame_id_to_rtp_timestamp_[256]; | 166 RtpTimestamp frame_id_to_rtp_timestamp_[256]; |
174 | 167 |
175 // NOTE: Weak pointers must be invalidated before all other member variables. | 168 // NOTE: Weak pointers must be invalidated before all other member variables. |
176 base::WeakPtrFactory<AudioReceiver> weak_factory_; | 169 base::WeakPtrFactory<AudioReceiver> weak_factory_; |
177 | 170 |
178 DISALLOW_COPY_AND_ASSIGN(AudioReceiver); | 171 DISALLOW_COPY_AND_ASSIGN(AudioReceiver); |
179 }; | 172 }; |
180 | 173 |
181 } // namespace cast | 174 } // namespace cast |
182 } // namespace media | 175 } // namespace media |
183 | 176 |
184 #endif // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_ | 177 #endif // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_ |
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