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Side by Side Diff: media/cast/audio_receiver/audio_decoder.cc

Issue 24586003: Be able to build cast_unittest and related targets in Chrome tree (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 7 years, 2 months ago
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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
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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_);
pwestin 2013/09/25 21:24:46 This is needed to avoid a memory leak; why did you
Alpha Left Google 2013/09/25 21:26:21 I can't find reference to ACM::Destroy() in the he
53 } 52 }
54 53
55 bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks, 54 bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks,
56 int desired_frequency, 55 int desired_frequency,
57 PcmAudioFrame* audio_frame, 56 PcmAudioFrame* audio_frame,
58 uint32* rtp_timestamp) { 57 uint32* rtp_timestamp) {
59 if (!have_received_packets_) return false; 58 if (!have_received_packets_) return false;
60 59
61 for (int i = 0; i < number_of_10ms_blocks; ++i) { 60 for (int i = 0; i < number_of_10ms_blocks; ++i) {
62 webrtc::AudioFrame webrtc_audio_frame; 61 webrtc::AudioFrame webrtc_audio_frame;
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89 } 88 }
90 89
91 void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data, 90 void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data,
92 int payload_size, 91 int payload_size,
93 const RtpCastHeader& rtp_header) { 92 const RtpCastHeader& rtp_header) {
94 have_received_packets_ = true; 93 have_received_packets_ = true;
95 audio_decoder_->IncomingPacket(payload_data, payload_size, rtp_header.webrtc); 94 audio_decoder_->IncomingPacket(payload_data, payload_size, rtp_header.webrtc);
96 } 95 }
97 96
98 } // namespace cast 97 } // namespace cast
99 } // namespace media 98 } // namespace media
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