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_SENDER_AUDIO_SENDER_H_
6 #define MEDIA_CAST_SENDER_AUDIO_SENDER_H_
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/sender/frame_sender.h"
24 // Not thread safe. Only called from the main cast thread.
25 // This class owns all objects related to sending audio, objects that create RTP
26 // packets, congestion control, audio encoder, parsing and sending of
28 // Additionally it posts a bunch of delayed tasks to the main thread for various
30 class AudioSender : public FrameSender,
31 public base::NonThreadSafe,
32 public base::SupportsWeakPtr<AudioSender> {
34 AudioSender(scoped_refptr<CastEnvironment> cast_environment,
35 const AudioSenderConfig& audio_config,
36 CastTransportSender* const transport_sender);
38 virtual ~AudioSender();
40 CastInitializationStatus InitializationResult() const {
41 return cast_initialization_status_;
44 // Note: It is not guaranteed that |audio_frame| will actually be encoded and
45 // sent, if AudioSender detects too many frames in flight. Therefore, clients
46 // should be careful about the rate at which this method is called.
48 // Note: It is invalid to call this method if InitializationResult() returns
49 // anything but STATUS_AUDIO_INITIALIZED.
50 void InsertAudio(scoped_ptr<AudioBus> audio_bus,
51 const base::TimeTicks& recorded_time);
54 // Protected for testability.
55 void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback);
58 // Schedule and execute periodic checks for re-sending packets. If no
59 // acknowledgements have been received for "too long," AudioSender will
60 // speculatively re-send certain packets of an unacked frame to kick-start
61 // re-transmission. This is a last resort tactic to prevent the session from
62 // getting stuck after a long outage.
63 void ScheduleNextResendCheck();
65 void ResendForKickstart();
67 // Returns true if there are too many frames in flight, as defined by the
68 // configured target playout delay plus simple logic. When this is true,
69 // InsertAudio() will silenty drop frames instead of sending them to the audio
71 bool AreTooManyFramesInFlight() const;
73 // Called by the |audio_encoder_| with the next EncodedFrame to send.
74 void SendEncodedAudioFrame(scoped_ptr<EncodedFrame> audio_frame);
76 // The total amount of time between a frame's capture/recording on the sender
77 // and its playback on the receiver (i.e., shown to a user). This is fixed as
78 // a value large enough to give the system sufficient time to encode,
79 // transmit/retransmit, receive, decode, and render; given its run-time
80 // environment (sender/receiver hardware performance, network conditions,
82 const base::TimeDelta target_playout_delay_;
84 // Maximum number of outstanding frames before the encoding and sending of
85 // new frames shall halt.
86 const int max_unacked_frames_;
88 // Encodes AudioBuses into EncodedFrames.
89 scoped_ptr<AudioEncoder> audio_encoder_;
90 const int configured_encoder_bitrate_;
92 // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
93 // frame) at the start of the session. Once a threshold is reached, RTCP
94 // reports are instead sent at the configured interval + random drift.
95 int num_aggressive_rtcp_reports_sent_;
97 // This is "null" until the first frame is sent. Thereafter, this tracks the
98 // last time any frame was sent or re-sent.
99 base::TimeTicks last_send_time_;
101 // The ID of the last frame sent. Logic throughout AudioSender assumes this
102 // can safely wrap-around. This member is invalid until
103 // |!last_send_time_.is_null()|.
104 uint32 last_sent_frame_id_;
106 // The ID of the latest (not necessarily the last) frame that has been
107 // acknowledged. Logic throughout AudioSender assumes this can safely
108 // wrap-around. This member is invalid until |!last_send_time_.is_null()|.
109 uint32 latest_acked_frame_id_;
111 // Counts the number of duplicate ACK that are being received. When this
112 // number reaches a threshold, the sender will take this as a sign that the
113 // receiver hasn't yet received the first packet of the next frame. In this
114 // case, AudioSender will trigger a re-send of the next frame.
115 int duplicate_ack_counter_;
117 // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED.
118 CastInitializationStatus cast_initialization_status_;
120 // This is a "good enough" mapping for finding the RTP timestamp associated
121 // with a video frame. The key is the lowest 8 bits of frame id (which is
122 // what is sent via RTCP). This map is used for logging purposes.
123 RtpTimestamp frame_id_to_rtp_timestamp_[256];
125 // NOTE: Weak pointers must be invalidated before all other member variables.
126 base::WeakPtrFactory<AudioSender> weak_factory_;
128 DISALLOW_COPY_AND_ASSIGN(AudioSender);
134 #endif // MEDIA_CAST_SENDER_AUDIO_SENDER_H_