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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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/audio/mac/audio_output_mac.h" | 5 #include "media/audio/mac/audio_output_mac.h" |
6 | 6 |
7 #include <CoreServices/CoreServices.h> | 7 #include <CoreServices/CoreServices.h> |
8 | 8 |
9 #include "base/basictypes.h" | 9 #include "base/basictypes.h" |
10 #include "base/debug/trace_event.h" | 10 #include "base/debug/trace_event.h" |
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41 enum { | 41 enum { |
42 kAudioQueueErr_EnqueueDuringReset = -66632 | 42 kAudioQueueErr_EnqueueDuringReset = -66632 |
43 }; | 43 }; |
44 #endif | 44 #endif |
45 | 45 |
46 PCMQueueOutAudioOutputStream::PCMQueueOutAudioOutputStream( | 46 PCMQueueOutAudioOutputStream::PCMQueueOutAudioOutputStream( |
47 AudioManagerMac* manager, const AudioParameters& params) | 47 AudioManagerMac* manager, const AudioParameters& params) |
48 : audio_queue_(NULL), | 48 : audio_queue_(NULL), |
49 source_(NULL), | 49 source_(NULL), |
50 manager_(manager), | 50 manager_(manager), |
51 packet_size_(params.GetPacketSize()), | 51 packet_size_(params.GetBytesPerBuffer()), |
52 silence_bytes_(0), | 52 silence_bytes_(0), |
53 volume_(1), | 53 volume_(1), |
54 pending_bytes_(0), | 54 pending_bytes_(0), |
55 num_source_channels_(params.channels), | 55 num_source_channels_(params.channels()), |
56 source_layout_(params.channel_layout), | 56 source_layout_(params.channel_layout()), |
57 num_core_channels_(0), | 57 num_core_channels_(0), |
58 should_swizzle_(false), | 58 should_swizzle_(false), |
59 should_down_mix_(false) { | 59 should_down_mix_(false) { |
60 // We must have a manager. | 60 // We must have a manager. |
61 DCHECK(manager_); | 61 DCHECK(manager_); |
62 // A frame is one sample across all channels. In interleaved audio the per | 62 // A frame is one sample across all channels. In interleaved audio the per |
63 // frame fields identify the set of n |channels|. In uncompressed audio, a | 63 // frame fields identify the set of n |channels|. In uncompressed audio, a |
64 // packet is always one frame. | 64 // packet is always one frame. |
65 format_.mSampleRate = params.sample_rate; | 65 format_.mSampleRate = params.sample_rate(); |
66 format_.mFormatID = kAudioFormatLinearPCM; | 66 format_.mFormatID = kAudioFormatLinearPCM; |
67 format_.mFormatFlags = kLinearPCMFormatFlagIsPacked; | 67 format_.mFormatFlags = kLinearPCMFormatFlagIsPacked; |
68 format_.mBitsPerChannel = params.bits_per_sample; | 68 format_.mBitsPerChannel = params.bits_per_sample(); |
69 format_.mChannelsPerFrame = params.channels; | 69 format_.mChannelsPerFrame = params.channels(); |
70 format_.mFramesPerPacket = 1; | 70 format_.mFramesPerPacket = 1; |
71 format_.mBytesPerPacket = (format_.mBitsPerChannel * params.channels) / 8; | 71 format_.mBytesPerPacket = (format_.mBitsPerChannel * params.channels()) / 8; |
72 format_.mBytesPerFrame = format_.mBytesPerPacket; | 72 format_.mBytesPerFrame = format_.mBytesPerPacket; |
73 format_.mReserved = 0; | 73 format_.mReserved = 0; |
74 | 74 |
75 memset(buffer_, 0, sizeof(buffer_)); | 75 memset(buffer_, 0, sizeof(buffer_)); |
76 memset(core_channel_orderings_, 0, sizeof(core_channel_orderings_)); | 76 memset(core_channel_orderings_, 0, sizeof(core_channel_orderings_)); |
77 memset(channel_remap_, 0, sizeof(channel_remap_)); | 77 memset(channel_remap_, 0, sizeof(channel_remap_)); |
78 | 78 |
79 if (params.bits_per_sample > 8) { | 79 if (params.bits_per_sample() > 8) { |
80 format_.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger; | 80 format_.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger; |
81 } | 81 } |
82 | 82 |
83 // Silence buffer has a duration of 6ms to simulate the behavior of Windows. | 83 // Silence buffer has a duration of 6ms to simulate the behavior of Windows. |
84 // This value is choosen by experiments and macs cannot keep up with | 84 // This value is choosen by experiments and macs cannot keep up with |
85 // anything less than 6ms. | 85 // anything less than 6ms. |
86 silence_bytes_ = format_.mBytesPerFrame * params.sample_rate * 6 / 1000; | 86 silence_bytes_ = format_.mBytesPerFrame * params.sample_rate() * 6 / 1000; |
87 } | 87 } |
88 | 88 |
89 PCMQueueOutAudioOutputStream::~PCMQueueOutAudioOutputStream() { | 89 PCMQueueOutAudioOutputStream::~PCMQueueOutAudioOutputStream() { |
90 } | 90 } |
91 | 91 |
92 void PCMQueueOutAudioOutputStream::HandleError(OSStatus err) { | 92 void PCMQueueOutAudioOutputStream::HandleError(OSStatus err) { |
93 // source_ can be set to NULL from another thread. We need to cache its | 93 // source_ can be set to NULL from another thread. We need to cache its |
94 // pointer while we operate here. Note that does not mean that the source | 94 // pointer while we operate here. Note that does not mean that the source |
95 // has been destroyed. | 95 // has been destroyed. |
96 AudioSourceCallback* source = GetSource(); | 96 AudioSourceCallback* source = GetSource(); |
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520 void PCMQueueOutAudioOutputStream::SetSource(AudioSourceCallback* source) { | 520 void PCMQueueOutAudioOutputStream::SetSource(AudioSourceCallback* source) { |
521 base::AutoLock lock(source_lock_); | 521 base::AutoLock lock(source_lock_); |
522 source_ = source; | 522 source_ = source; |
523 } | 523 } |
524 | 524 |
525 AudioOutputStream::AudioSourceCallback* | 525 AudioOutputStream::AudioSourceCallback* |
526 PCMQueueOutAudioOutputStream::GetSource() { | 526 PCMQueueOutAudioOutputStream::GetSource() { |
527 base::AutoLock lock(source_lock_); | 527 base::AutoLock lock(source_lock_); |
528 return source_; | 528 return source_; |
529 } | 529 } |
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