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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 "crypto/symmetric_key.h" |
10 #include "media/cast/audio_sender/audio_encoder.h" | 11 #include "media/cast/audio_sender/audio_encoder.h" |
11 #include "media/cast/rtcp/rtcp.h" | 12 #include "media/cast/rtcp/rtcp.h" |
12 #include "media/cast/rtp_sender/rtp_sender.h" | 13 #include "media/cast/rtp_sender/rtp_sender.h" |
13 | 14 |
14 namespace media { | 15 namespace media { |
15 namespace cast { | 16 namespace cast { |
16 | 17 |
17 const int64 kMinSchedulingDelayMs = 1; | 18 const int64 kMinSchedulingDelayMs = 1; |
18 | 19 |
19 class LocalRtcpAudioSenderFeedback : public RtcpSenderFeedback { | 20 class LocalRtcpAudioSenderFeedback : public RtcpSenderFeedback { |
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86 rtcp_feedback_.get(), | 87 rtcp_feedback_.get(), |
87 paced_packet_sender, | 88 paced_packet_sender, |
88 rtp_audio_sender_statistics_.get(), | 89 rtp_audio_sender_statistics_.get(), |
89 NULL, | 90 NULL, |
90 audio_config.rtcp_mode, | 91 audio_config.rtcp_mode, |
91 base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval), | 92 base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval), |
92 true, | 93 true, |
93 audio_config.sender_ssrc, | 94 audio_config.sender_ssrc, |
94 audio_config.rtcp_c_name), | 95 audio_config.rtcp_c_name), |
95 weak_factory_(this) { | 96 weak_factory_(this) { |
| 97 if (audio_config.aes_iv_mask.size() == kAesKeySize && |
| 98 audio_config.aes_key.size() == kAesKeySize) { |
| 99 iv_mask_ = audio_config.aes_iv_mask; |
| 100 crypto::SymmetricKey* key = crypto::SymmetricKey::Import( |
| 101 crypto::SymmetricKey::AES, audio_config.aes_key); |
| 102 encryptor_.reset(new crypto::Encryptor()); |
| 103 encryptor_->Init(key, crypto::Encryptor::CTR, std::string()); |
| 104 } else if (audio_config.aes_iv_mask.size() != 0 || |
| 105 audio_config.aes_key.size() != 0) { |
| 106 DCHECK(false) << "Invalid crypto configuration"; |
| 107 } |
| 108 |
96 rtcp_.SetRemoteSSRC(audio_config.incoming_feedback_ssrc); | 109 rtcp_.SetRemoteSSRC(audio_config.incoming_feedback_ssrc); |
97 | 110 |
98 if (!audio_config.use_external_encoder) { | 111 if (!audio_config.use_external_encoder) { |
99 audio_encoder_ = new AudioEncoder(cast_environment, audio_config); | 112 audio_encoder_ = new AudioEncoder(cast_environment, audio_config); |
100 } | 113 } |
101 ScheduleNextRtcpReport(); | 114 ScheduleNextRtcpReport(); |
102 } | 115 } |
103 | 116 |
104 AudioSender::~AudioSender() {} | 117 AudioSender::~AudioSender() {} |
105 | 118 |
106 void AudioSender::InsertRawAudioFrame( | 119 void AudioSender::InsertRawAudioFrame( |
107 const PcmAudioFrame* audio_frame, | 120 const PcmAudioFrame* audio_frame, |
108 const base::TimeTicks& recorded_time, | 121 const base::TimeTicks& recorded_time, |
109 const base::Closure callback) { | 122 const base::Closure callback) { |
110 DCHECK(audio_encoder_.get()) << "Invalid internal state"; | 123 DCHECK(audio_encoder_.get()) << "Invalid internal state"; |
111 | 124 |
112 audio_encoder_->InsertRawAudioFrame(audio_frame, recorded_time, | 125 audio_encoder_->InsertRawAudioFrame(audio_frame, recorded_time, |
113 base::Bind(&AudioSender::SendEncodedAudioFrame, | 126 base::Bind(&AudioSender::SendEncodedAudioFrame, |
114 weak_factory_.GetWeakPtr()), | 127 weak_factory_.GetWeakPtr()), |
115 callback); | 128 callback); |
116 } | 129 } |
117 | 130 |
118 void AudioSender::InsertCodedAudioFrame(const EncodedAudioFrame* audio_frame, | 131 void AudioSender::InsertCodedAudioFrame(const EncodedAudioFrame* audio_frame, |
119 const base::TimeTicks& recorded_time, | 132 const base::TimeTicks& recorded_time, |
120 const base::Closure callback) { | 133 const base::Closure callback) { |
121 DCHECK(audio_encoder_.get() == NULL) << "Invalid internal state"; | 134 DCHECK(audio_encoder_.get() == NULL) << "Invalid internal state"; |
122 rtp_sender_.IncomingEncodedAudioFrame(audio_frame, recorded_time); | 135 |
| 136 if (encryptor_) { |
| 137 EncodedAudioFrame encrypted_frame; |
| 138 if (!EncryptAudioFrame(*audio_frame, &encrypted_frame)) { |
| 139 NOTREACHED() << "Encryption error"; |
| 140 return; |
| 141 } |
| 142 rtp_sender_.IncomingEncodedAudioFrame(&encrypted_frame, recorded_time); |
| 143 } else { |
| 144 rtp_sender_.IncomingEncodedAudioFrame(audio_frame, recorded_time); |
| 145 } |
123 callback.Run(); | 146 callback.Run(); |
124 } | 147 } |
125 | 148 |
126 void AudioSender::SendEncodedAudioFrame( | 149 void AudioSender::SendEncodedAudioFrame( |
127 scoped_ptr<EncodedAudioFrame> audio_frame, | 150 scoped_ptr<EncodedAudioFrame> audio_frame, |
128 const base::TimeTicks& recorded_time) { | 151 const base::TimeTicks& recorded_time) { |
129 rtp_sender_.IncomingEncodedAudioFrame(audio_frame.get(), recorded_time); | 152 if (encryptor_) { |
| 153 EncodedAudioFrame encrypted_frame; |
| 154 if (!EncryptAudioFrame(*audio_frame.get(), &encrypted_frame)) { |
| 155 NOTREACHED() << "Encryption error"; |
| 156 return; |
| 157 } |
| 158 rtp_sender_.IncomingEncodedAudioFrame(&encrypted_frame, recorded_time); |
| 159 } else { |
| 160 rtp_sender_.IncomingEncodedAudioFrame(audio_frame.get(), recorded_time); |
| 161 } |
| 162 } |
| 163 |
| 164 bool AudioSender::EncryptAudioFrame(const EncodedAudioFrame& audio_frame, |
| 165 EncodedAudioFrame* encrypted_frame) { |
| 166 DCHECK(encryptor_) << "Invalid state"; |
| 167 |
| 168 if (!encryptor_->SetCounter(GetAesNonce(audio_frame.frame_id, iv_mask_))) { |
| 169 NOTREACHED(); |
| 170 return false; |
| 171 } |
| 172 if (!encryptor_->Encrypt(audio_frame.data, &encrypted_frame->data)) { |
| 173 VLOG(0) << "Encrypt error"; |
| 174 return false; |
| 175 } |
| 176 encrypted_frame->codec = audio_frame.codec; |
| 177 encrypted_frame->frame_id = audio_frame.frame_id; |
| 178 encrypted_frame->samples = audio_frame.samples; |
| 179 return true; |
130 } | 180 } |
131 | 181 |
132 void AudioSender::ResendPackets( | 182 void AudioSender::ResendPackets( |
133 const MissingFramesAndPacketsMap& missing_frames_and_packets) { | 183 const MissingFramesAndPacketsMap& missing_frames_and_packets) { |
134 rtp_sender_.ResendPackets(missing_frames_and_packets); | 184 rtp_sender_.ResendPackets(missing_frames_and_packets); |
135 } | 185 } |
136 | 186 |
137 void AudioSender::IncomingRtcpPacket(const uint8* packet, size_t length, | 187 void AudioSender::IncomingRtcpPacket(const uint8* packet, size_t length, |
138 const base::Closure callback) { | 188 const base::Closure callback) { |
139 rtcp_.IncomingRtcpPacket(packet, length); | 189 rtcp_.IncomingRtcpPacket(packet, length); |
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152 time_to_next); | 202 time_to_next); |
153 } | 203 } |
154 | 204 |
155 void AudioSender::SendRtcpReport() { | 205 void AudioSender::SendRtcpReport() { |
156 rtcp_.SendRtcpReport(incoming_feedback_ssrc_); | 206 rtcp_.SendRtcpReport(incoming_feedback_ssrc_); |
157 ScheduleNextRtcpReport(); | 207 ScheduleNextRtcpReport(); |
158 } | 208 } |
159 | 209 |
160 } // namespace cast | 210 } // namespace cast |
161 } // namespace media | 211 } // namespace media |
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