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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 "base/logging.h" | 5 #include "base/logging.h" |
6 #include "media/cast/audio_receiver/audio_decoder.h" | 6 #include "media/cast/audio_receiver/audio_decoder.h" |
7 | 7 |
8 #include "third_party/webrtc/modules/audio_coding/main/interface/audio_coding_mo
dule.h" | 8 #include "third_party/webrtc/modules/audio_coding/main/interface/audio_coding_mo
dule.h" |
9 #include "third_party/webrtc/modules/interface/module_common_types.h" | 9 #include "third_party/webrtc/modules/interface/module_common_types.h" |
10 | 10 |
11 namespace media { | 11 namespace media { |
12 namespace cast { | 12 namespace cast { |
13 | 13 |
14 AudioDecoder::AudioDecoder(scoped_refptr<CastThread> cast_thread, | 14 AudioDecoder::AudioDecoder(scoped_refptr<CastThread> cast_thread, |
15 const AudioReceiverConfig& audio_config) | 15 const AudioReceiverConfig& audio_config) |
16 : audio_decoder_(webrtc::AudioCodingModule::Create(0)), | 16 : cast_thread_(cast_thread), |
17 have_received_packets_(false), | 17 have_received_packets_(false) { |
18 cast_thread_(cast_thread) { | 18 audio_decoder_ = webrtc::AudioCodingModule::Create(0); |
19 audio_decoder_->InitializeReceiver(); | 19 audio_decoder_->InitializeReceiver(); |
20 | 20 |
21 webrtc::CodecInst receive_codec; | 21 webrtc::CodecInst receive_codec; |
22 switch (audio_config.codec) { | 22 switch (audio_config.codec) { |
23 case kPcm16: | 23 case kPcm16: |
24 receive_codec.pltype = audio_config.rtp_payload_type; | 24 receive_codec.pltype = audio_config.rtp_payload_type; |
25 strncpy(receive_codec.plname, "L16", 4); | 25 strncpy(receive_codec.plname, "L16", 4); |
26 receive_codec.plfreq = audio_config.frequency; | 26 receive_codec.plfreq = audio_config.frequency; |
27 receive_codec.pacsize = -1; | 27 receive_codec.pacsize = -1; |
28 receive_codec.channels = audio_config.channels; | 28 receive_codec.channels = audio_config.channels; |
(...skipping 13 matching lines...) Expand all Loading... |
42 } | 42 } |
43 if (audio_decoder_->RegisterReceiveCodec(receive_codec) != 0) { | 43 if (audio_decoder_->RegisterReceiveCodec(receive_codec) != 0) { |
44 DCHECK(false) << "Failed to register receive codec"; | 44 DCHECK(false) << "Failed to register receive codec"; |
45 } | 45 } |
46 | 46 |
47 audio_decoder_->SetMaximumPlayoutDelay(audio_config.rtp_max_delay_ms); | 47 audio_decoder_->SetMaximumPlayoutDelay(audio_config.rtp_max_delay_ms); |
48 audio_decoder_->SetPlayoutMode(webrtc::streaming); | 48 audio_decoder_->SetPlayoutMode(webrtc::streaming); |
49 } | 49 } |
50 | 50 |
51 AudioDecoder::~AudioDecoder() { | 51 AudioDecoder::~AudioDecoder() { |
| 52 webrtc::AudioCodingModule::Destroy(audio_decoder_); |
52 } | 53 } |
53 | 54 |
54 bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks, | 55 bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks, |
55 int desired_frequency, | 56 int desired_frequency, |
56 PcmAudioFrame* audio_frame, | 57 PcmAudioFrame* audio_frame, |
57 uint32* rtp_timestamp) { | 58 uint32* rtp_timestamp) { |
58 DCHECK(cast_thread_->CurrentlyOn(CastThread::AUDIO_DECODER)); | 59 DCHECK(cast_thread_->CurrentlyOn(CastThread::AUDIO_DECODER)); |
59 if (!have_received_packets_) return false; | 60 if (!have_received_packets_) return false; |
60 | 61 |
61 for (int i = 0; i < number_of_10ms_blocks; ++i) { | 62 for (int i = 0; i < number_of_10ms_blocks; ++i) { |
(...skipping 27 matching lines...) Expand all Loading... |
89 } | 90 } |
90 | 91 |
91 void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data, | 92 void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data, |
92 int payload_size, | 93 int payload_size, |
93 const RtpCastHeader& rtp_header) { | 94 const RtpCastHeader& rtp_header) { |
94 have_received_packets_ = true; | 95 have_received_packets_ = true; |
95 audio_decoder_->IncomingPacket(payload_data, payload_size, rtp_header.webrtc); | 96 audio_decoder_->IncomingPacket(payload_data, payload_size, rtp_header.webrtc); |
96 } | 97 } |
97 | 98 |
98 } // namespace cast | 99 } // namespace cast |
99 } // namespace media | 100 } // namespace media |
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