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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 #include "media/cast/audio_sender/audio_sender.h" | 5 #include "media/cast/audio_sender/audio_sender.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/logging.h" | 8 #include "base/logging.h" |
9 #include "base/message_loop/message_loop.h" | 9 #include "base/message_loop/message_loop.h" |
10 #include "media/cast/audio_sender/audio_encoder.h" | 10 #include "media/cast/audio_sender/audio_encoder.h" |
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34 | 34 |
35 DISALLOW_IMPLICIT_CONSTRUCTORS(LocalRtcpAudioSenderFeedback); | 35 DISALLOW_IMPLICIT_CONSTRUCTORS(LocalRtcpAudioSenderFeedback); |
36 }; | 36 }; |
37 | 37 |
38 // TODO(mikhal): Reduce heap allocation when not needed. | 38 // TODO(mikhal): Reduce heap allocation when not needed. |
39 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment, | 39 AudioSender::AudioSender(scoped_refptr<CastEnvironment> cast_environment, |
40 const AudioSenderConfig& audio_config, | 40 const AudioSenderConfig& audio_config, |
41 transport::CastTransportSender* const transport_sender) | 41 transport::CastTransportSender* const transport_sender) |
42 : cast_environment_(cast_environment), | 42 : cast_environment_(cast_environment), |
43 transport_sender_(transport_sender), | 43 transport_sender_(transport_sender), |
44 rtp_stats_(audio_config.frequency), | 44 rtp_timestamp_helper_(audio_config.frequency), |
45 rtcp_feedback_(new LocalRtcpAudioSenderFeedback(this)), | 45 rtcp_feedback_(new LocalRtcpAudioSenderFeedback(this)), |
46 rtcp_(cast_environment, | 46 rtcp_(cast_environment, |
47 rtcp_feedback_.get(), | 47 rtcp_feedback_.get(), |
48 transport_sender_, | 48 transport_sender_, |
49 NULL, // paced sender. | 49 NULL, // paced sender. |
50 NULL, | 50 NULL, |
51 audio_config.rtcp_mode, | 51 audio_config.rtcp_mode, |
52 base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval), | 52 base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval), |
53 audio_config.rtp_config.ssrc, | 53 audio_config.rtp_config.ssrc, |
54 audio_config.incoming_feedback_ssrc, | 54 audio_config.incoming_feedback_ssrc, |
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68 } | 68 } |
69 | 69 |
70 media::cast::transport::CastTransportAudioConfig transport_config; | 70 media::cast::transport::CastTransportAudioConfig transport_config; |
71 transport_config.codec = audio_config.codec; | 71 transport_config.codec = audio_config.codec; |
72 transport_config.rtp.config = audio_config.rtp_config; | 72 transport_config.rtp.config = audio_config.rtp_config; |
73 transport_config.frequency = audio_config.frequency; | 73 transport_config.frequency = audio_config.frequency; |
74 transport_config.channels = audio_config.channels; | 74 transport_config.channels = audio_config.channels; |
75 transport_config.rtp.max_outstanding_frames = | 75 transport_config.rtp.max_outstanding_frames = |
76 audio_config.rtp_config.max_delay_ms / 100 + 1; | 76 audio_config.rtp_config.max_delay_ms / 100 + 1; |
77 transport_sender_->InitializeAudio(transport_config); | 77 transport_sender_->InitializeAudio(transport_config); |
78 | |
79 transport_sender_->SubscribeAudioRtpStatsCallback( | |
80 base::Bind(&AudioSender::StoreStatistics, weak_factory_.GetWeakPtr())); | |
81 } | 78 } |
82 | 79 |
83 AudioSender::~AudioSender() {} | 80 AudioSender::~AudioSender() {} |
84 | 81 |
85 void AudioSender::InitializeTimers() { | 82 void AudioSender::InitializeTimers() { |
86 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 83 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
87 if (!timers_initialized_) { | 84 if (!timers_initialized_) { |
88 timers_initialized_ = true; | 85 timers_initialized_ = true; |
89 ScheduleNextRtcpReport(); | 86 ScheduleNextRtcpReport(); |
90 } | 87 } |
91 } | 88 } |
92 | 89 |
93 void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus, | 90 void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus, |
94 const base::TimeTicks& recorded_time) { | 91 const base::TimeTicks& recorded_time) { |
95 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 92 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
96 DCHECK(audio_encoder_.get()) << "Invalid internal state"; | 93 DCHECK(audio_encoder_.get()) << "Invalid internal state"; |
97 audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time); | 94 audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time); |
98 } | 95 } |
99 | 96 |
100 void AudioSender::SendEncodedAudioFrame( | 97 void AudioSender::SendEncodedAudioFrame( |
101 scoped_ptr<transport::EncodedAudioFrame> audio_frame, | 98 scoped_ptr<transport::EncodedAudioFrame> audio_frame, |
102 const base::TimeTicks& recorded_time) { | 99 const base::TimeTicks& recorded_time) { |
103 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 100 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
| 101 rtp_timestamp_helper_.StoreLatestTime(recorded_time, |
| 102 audio_frame->rtp_timestamp); |
104 InitializeTimers(); | 103 InitializeTimers(); |
105 transport_sender_->InsertCodedAudioFrame(audio_frame.get(), recorded_time); | 104 transport_sender_->InsertCodedAudioFrame(audio_frame.get(), recorded_time); |
106 } | 105 } |
107 | 106 |
108 void AudioSender::ResendPackets( | 107 void AudioSender::ResendPackets( |
109 const MissingFramesAndPacketsMap& missing_frames_and_packets) { | 108 const MissingFramesAndPacketsMap& missing_frames_and_packets) { |
110 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 109 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
111 transport_sender_->ResendPackets(true, missing_frames_and_packets); | 110 transport_sender_->ResendPackets(true, missing_frames_and_packets); |
112 } | 111 } |
113 | 112 |
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124 time_to_next = std::max( | 123 time_to_next = std::max( |
125 time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs)); | 124 time_to_next, base::TimeDelta::FromMilliseconds(kMinSchedulingDelayMs)); |
126 | 125 |
127 cast_environment_->PostDelayedTask( | 126 cast_environment_->PostDelayedTask( |
128 CastEnvironment::MAIN, | 127 CastEnvironment::MAIN, |
129 FROM_HERE, | 128 FROM_HERE, |
130 base::Bind(&AudioSender::SendRtcpReport, weak_factory_.GetWeakPtr()), | 129 base::Bind(&AudioSender::SendRtcpReport, weak_factory_.GetWeakPtr()), |
131 time_to_next); | 130 time_to_next); |
132 } | 131 } |
133 | 132 |
134 void AudioSender::StoreStatistics( | |
135 const transport::RtcpSenderInfo& sender_info, | |
136 base::TimeTicks time_sent, | |
137 uint32 rtp_timestamp) { | |
138 rtp_stats_.Store(sender_info, time_sent, rtp_timestamp); | |
139 } | |
140 | |
141 void AudioSender::SendRtcpReport() { | 133 void AudioSender::SendRtcpReport() { |
142 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); | 134 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
143 // We don't send audio logging messages since all captured audio frames will | 135 const base::TimeTicks now = cast_environment_->Clock()->NowTicks(); |
144 // be sent. | 136 uint32 now_as_rtp_timestamp = 0; |
145 rtp_stats_.UpdateInfo(cast_environment_->Clock()->NowTicks()); | 137 if (rtp_timestamp_helper_.GetCurrentTimeAsRtpTimestamp( |
146 rtcp_.SendRtcpFromRtpSender(rtp_stats_.sender_info()); | 138 now, &now_as_rtp_timestamp)) { |
| 139 rtcp_.SendRtcpFromRtpSender(now, now_as_rtp_timestamp); |
| 140 } |
147 ScheduleNextRtcpReport(); | 141 ScheduleNextRtcpReport(); |
148 } | 142 } |
149 | 143 |
150 } // namespace cast | 144 } // namespace cast |
151 } // namespace media | 145 } // namespace media |
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