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[chromium-blink-merge.git] / media / cast / audio_receiver / audio_receiver.h
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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.
5 #ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
6 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
8 #include "base/basictypes.h"
9 #include "base/callback.h"
10 #include "base/macros.h"
11 #include "base/memory/ref_counted.h"
12 #include "base/memory/scoped_ptr.h"
13 #include "base/memory/weak_ptr.h"
14 #include "base/threading/non_thread_safe.h"
15 #include "base/time/tick_clock.h"
16 #include "base/time/time.h"
17 #include "media/cast/cast_config.h"
18 #include "media/cast/cast_environment.h"
19 #include "media/cast/cast_receiver.h"
20 #include "media/cast/framer/framer.h"
21 #include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
22 #include "media/cast/rtcp/rtcp.h"
23 #include "media/cast/rtp_receiver/rtp_receiver.h"
24 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
25 #include "media/cast/transport/utility/transport_encryption_handler.h"
27 namespace media {
28 namespace cast {
30 class AudioDecoder;
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.)
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
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
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
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
47 // 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
49 // decoder. Each request for a frame includes a callback which AudioReceiver
50 // guarantees will be called at some point in the future unless the
51 // AudioReceiver is destroyed. Clients should generally limit the number of
52 // outstanding requests (perhaps to just one or two).
54 // This class is not thread safe. Should only be called from the Main cast
55 // thread.
56 class AudioReceiver : public RtpReceiver,
57 public RtpPayloadFeedback,
58 public base::NonThreadSafe,
59 public base::SupportsWeakPtr<AudioReceiver> {
60 public:
61 AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
62 const AudioReceiverConfig& audio_config,
63 transport::PacedPacketSender* const packet_sender);
65 virtual ~AudioReceiver();
67 // Request a decoded audio frame. The audio signal data returned in the
68 // callback will have the sampling rate and number of channels as requested in
69 // the configuration that was passed to the ctor.
71 // The given |callback| is guaranteed to be run at some point in the future,
72 // even if to respond with NULL at shutdown time.
73 void GetRawAudioFrame(const AudioFrameDecodedCallback& callback);
75 // Request an encoded audio frame.
77 // The given |callback| is guaranteed to be run at some point in the future,
78 // even if to respond with NULL at shutdown time.
79 void GetEncodedAudioFrame(const FrameEncodedCallback& callback);
81 // Deliver another packet, possibly a duplicate, and possibly out-of-order.
82 void IncomingPacket(scoped_ptr<Packet> packet);
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);
88 protected:
89 friend class AudioReceiverTest; // Invokes OnReceivedPayloadData().
91 virtual void OnReceivedPayloadData(const uint8* payload_data,
92 size_t payload_size,
93 const RtpCastHeader& rtp_header) OVERRIDE;
95 // RtpPayloadFeedback implementation.
96 virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE;
98 private:
99 // Processes ready-to-consume packets from |framer_|, decrypting each packet's
100 // 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
102 // the future to wait for missing/incomplete frames.
103 void EmitAvailableEncodedFrames();
105 // Clears the |is_waiting_for_consecutive_frame_| flag and invokes
106 // EmitAvailableEncodedFrames().
107 void EmitAvailableEncodedFramesAfterWaiting();
109 // Feeds an EncodedFrame into |audio_decoder_|. GetRawAudioFrame() uses this
110 // as a callback for GetEncodedAudioFrame().
111 void DecodeEncodedAudioFrame(
112 const AudioFrameDecodedCallback& callback,
113 scoped_ptr<transport::EncodedFrame> encoded_frame);
115 // Return the playout time based on the current time and rtp timestamp.
116 base::TimeTicks GetPlayoutTime(base::TimeTicks now, uint32 rtp_timestamp);
118 void InitializeTimers();
120 // Schedule the next RTCP report.
121 void ScheduleNextRtcpReport();
123 // Actually send the next RTCP report.
124 void SendNextRtcpReport();
126 // Schedule timing for the next cast message.
127 void ScheduleNextCastMessage();
129 // Actually send the next cast message.
130 void SendNextCastMessage();
132 // Receives an AudioBus from |audio_decoder_|, logs the event, and passes the
133 // data on by running the given |callback|. This method is static to ensure
134 // it can be called after an AudioReceiver instance is destroyed.
135 // DecodeEncodedAudioFrame() uses this as a callback for
136 // AudioDecoder::DecodeFrame().
137 static void EmitRawAudioFrame(
138 const scoped_refptr<CastEnvironment>& cast_environment,
139 const AudioFrameDecodedCallback& callback,
140 uint32 frame_id,
141 uint32 rtp_timestamp,
142 const base::TimeTicks& playout_time,
143 scoped_ptr<AudioBus> audio_bus,
144 bool is_continuous);
146 const scoped_refptr<CastEnvironment> cast_environment_;
148 // Subscribes to raw events.
149 // Processes raw audio events to be sent over to the cast sender via RTCP.
150 ReceiverRtcpEventSubscriber event_subscriber_;
152 const transport::AudioCodec codec_;
153 const int frequency_;
154 base::TimeDelta target_delay_delta_;
155 Framer framer_;
156 scoped_ptr<AudioDecoder> audio_decoder_;
157 Rtcp rtcp_;
158 base::TimeDelta time_offset_;
159 base::TimeTicks time_first_incoming_packet_;
160 uint32 first_incoming_rtp_timestamp_;
161 transport::TransportEncryptionHandler decryptor_;
163 // Outstanding callbacks to run to deliver on client requests for frames.
164 std::list<FrameEncodedCallback> frame_request_queue_;
166 // True while there's an outstanding task to re-invoke
167 // EmitAvailableEncodedFrames().
168 bool is_waiting_for_consecutive_frame_;
170 // This mapping allows us to log AUDIO_ACK_SENT as a frame event. In addition
171 // it allows the event to be transmitted via RTCP.
172 RtpTimestamp frame_id_to_rtp_timestamp_[256];
174 // NOTE: Weak pointers must be invalidated before all other member variables.
175 base::WeakPtrFactory<AudioReceiver> weak_factory_;
177 DISALLOW_COPY_AND_ASSIGN(AudioReceiver);
180 } // namespace cast
181 } // namespace media
183 #endif // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_