| OLD | NEW |
| 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 // Implementation of AudioInputStream for Mac OS X using the special AUHAL | 5 // Implementation of AudioInputStream for Mac OS X using the special AUHAL |
| 6 // input Audio Unit present in OS 10.4 and later. | 6 // input Audio Unit present in OS 10.4 and later. |
| 7 // The AUHAL input Audio Unit is for low-latency audio I/O. | 7 // The AUHAL input Audio Unit is for low-latency audio I/O. |
| 8 // | 8 // |
| 9 // Overview of operation: | 9 // Overview of operation: |
| 10 // | 10 // |
| (...skipping 27 matching lines...) Expand all Loading... |
| 38 | 38 |
| 39 #include <AudioUnit/AudioUnit.h> | 39 #include <AudioUnit/AudioUnit.h> |
| 40 #include <CoreAudio/CoreAudio.h> | 40 #include <CoreAudio/CoreAudio.h> |
| 41 | 41 |
| 42 #include "base/cancelable_callback.h" | 42 #include "base/cancelable_callback.h" |
| 43 #include "base/memory/scoped_ptr.h" | 43 #include "base/memory/scoped_ptr.h" |
| 44 #include "base/synchronization/lock.h" | 44 #include "base/synchronization/lock.h" |
| 45 #include "media/audio/agc_audio_stream.h" | 45 #include "media/audio/agc_audio_stream.h" |
| 46 #include "media/audio/audio_io.h" | 46 #include "media/audio/audio_io.h" |
| 47 #include "media/audio/audio_parameters.h" | 47 #include "media/audio/audio_parameters.h" |
| 48 #include "media/base/audio_block_fifo.h" | |
| 49 | 48 |
| 50 namespace media { | 49 namespace media { |
| 51 | 50 |
| 51 class AudioBlockFifo; |
| 52 class AudioBus; | 52 class AudioBus; |
| 53 class AudioManagerMac; | 53 class AudioManagerMac; |
| 54 class DataBuffer; | 54 class DataBuffer; |
| 55 | 55 |
| 56 class AUAudioInputStream : public AgcAudioStream<AudioInputStream> { | 56 class AUAudioInputStream : public AgcAudioStream<AudioInputStream> { |
| 57 public: | 57 public: |
| 58 // The ctor takes all the usual parameters, plus |manager| which is the | 58 // The ctor takes all the usual parameters, plus |manager| which is the |
| 59 // the audio manager who is creating this object. | 59 // the audio manager who is creating this object. |
| 60 AUAudioInputStream(AudioManagerMac* manager, | 60 AUAudioInputStream(AudioManagerMac* manager, |
| 61 const AudioParameters& input_params, | 61 const AudioParameters& input_params, |
| 62 AudioDeviceID audio_device_id); | 62 AudioDeviceID audio_device_id); |
| 63 // The dtor is typically called by the AudioManager only and it is usually | 63 // The dtor is typically called by the AudioManager only and it is usually |
| 64 // triggered by calling AudioInputStream::Close(). | 64 // triggered by calling AudioInputStream::Close(). |
| 65 virtual ~AUAudioInputStream(); | 65 virtual ~AUAudioInputStream(); |
| 66 | 66 |
| 67 // Implementation of AudioInputStream. | 67 // Implementation of AudioInputStream. |
| 68 virtual bool Open() OVERRIDE; | 68 virtual bool Open() OVERRIDE; |
| 69 virtual void Start(AudioInputCallback* callback) OVERRIDE; | 69 virtual void Start(AudioInputCallback* callback) OVERRIDE; |
| 70 virtual void Stop() OVERRIDE; | 70 virtual void Stop() OVERRIDE; |
| 71 virtual void Close() OVERRIDE; | 71 virtual void Close() OVERRIDE; |
| 72 virtual double GetMaxVolume() OVERRIDE; | 72 virtual double GetMaxVolume() OVERRIDE; |
| 73 virtual void SetVolume(double volume) OVERRIDE; | 73 virtual void SetVolume(double volume) OVERRIDE; |
| 74 virtual double GetVolume() OVERRIDE; | 74 virtual double GetVolume() OVERRIDE; |
| 75 | 75 |
| 76 // Returns the current hardware sample rate for the default input device. | 76 // Returns the current hardware sample rate for the default input device. |
| 77 MEDIA_EXPORT static int HardwareSampleRate(); | 77 MEDIA_EXPORT static int HardwareSampleRate(); |
| 78 | 78 |
| 79 bool started() const { return started_; } | 79 bool started() const { return started_; } |
| 80 AudioUnit audio_unit() { return audio_unit_; } | 80 AudioUnit audio_unit() { return audio_unit_; } |
| 81 AudioBufferList* audio_buffer_list() { return &audio_buffer_list_; } | 81 AudioBufferList* audio_buffer_list() { return audio_buffer_list_.get(); } |
| 82 | 82 |
| 83 private: | 83 private: |
| 84 // AudioOutputUnit callback. | 84 // AudioOutputUnit callback. |
| 85 static OSStatus InputProc(void* user_data, | 85 static OSStatus InputProc(void* user_data, |
| 86 AudioUnitRenderActionFlags* flags, | 86 AudioUnitRenderActionFlags* flags, |
| 87 const AudioTimeStamp* time_stamp, | 87 const AudioTimeStamp* time_stamp, |
| 88 UInt32 bus_number, | 88 UInt32 bus_number, |
| 89 UInt32 number_of_frames, | 89 UInt32 number_of_frames, |
| 90 AudioBufferList* io_data); | 90 AudioBufferList* io_data); |
| 91 | 91 |
| 92 // Pushes recorded data to consumer of the input audio stream. | 92 // Pushes recorded data to consumer of the input audio stream. |
| 93 OSStatus Provide(UInt32 number_of_frames, AudioBufferList* io_data, | 93 OSStatus Provide(UInt32 number_of_frames, |
| 94 AudioBufferList* io_data, |
| 94 const AudioTimeStamp* time_stamp); | 95 const AudioTimeStamp* time_stamp); |
| 95 | 96 |
| 96 // Gets the fixed capture hardware latency and store it during initialization. | 97 // Gets the fixed capture hardware latency and store it during initialization. |
| 97 // Returns 0 if not available. | 98 // Returns 0 if not available. |
| 98 double GetHardwareLatency(); | 99 double GetHardwareLatency(); |
| 99 | 100 |
| 100 // Gets the current capture delay value. | 101 // Gets the current capture delay value. |
| 101 double GetCaptureLatency(const AudioTimeStamp* input_time_stamp); | 102 double GetCaptureLatency(const AudioTimeStamp* input_time_stamp); |
| 102 | 103 |
| 103 // Gets the number of channels for a stream of audio data. | 104 // Gets the number of channels for a stream of audio data. |
| (...skipping 21 matching lines...) Expand all Loading... |
| 125 | 126 |
| 126 // The special Audio Unit called AUHAL, which allows us to pass audio data | 127 // The special Audio Unit called AUHAL, which allows us to pass audio data |
| 127 // directly from a microphone, through the HAL, and to our application. | 128 // directly from a microphone, through the HAL, and to our application. |
| 128 // The AUHAL also enables selection of non default devices. | 129 // The AUHAL also enables selection of non default devices. |
| 129 AudioUnit audio_unit_; | 130 AudioUnit audio_unit_; |
| 130 | 131 |
| 131 // The UID refers to the current input audio device. | 132 // The UID refers to the current input audio device. |
| 132 AudioDeviceID input_device_id_; | 133 AudioDeviceID input_device_id_; |
| 133 | 134 |
| 134 // Provides a mechanism for encapsulating one or more buffers of audio data. | 135 // Provides a mechanism for encapsulating one or more buffers of audio data. |
| 135 AudioBufferList audio_buffer_list_; | 136 scoped_ptr<AudioBufferList, base::FreeDeleter> audio_buffer_list_; |
| 136 | 137 |
| 137 // Temporary storage for recorded data. The InputProc() renders into this | 138 // Temporary storage for recorded data. The InputProc() renders into this |
| 138 // array as soon as a frame of the desired buffer size has been recorded. | 139 // array as soon as a frame of the desired buffer size has been recorded. |
| 139 scoped_ptr<uint8[]> audio_data_buffer_; | 140 scoped_ptr<float, base::AlignedFreeDeleter> audio_data_buffer_; |
| 140 | 141 |
| 141 // True after successfull Start(), false after successful Stop(). | 142 // True after successfull Start(), false after successful Stop(). |
| 142 bool started_; | 143 bool started_; |
| 143 | 144 |
| 144 // Fixed capture hardware latency in frames. | 145 // Fixed capture hardware latency in frames. |
| 145 double hardware_latency_frames_; | 146 double hardware_latency_frames_; |
| 146 | 147 |
| 147 // The number of channels in each frame of audio data, which is used | 148 // The number of channels in each frame of audio data, which is used |
| 148 // when querying the volume of each channel. | 149 // when querying the volume of each channel. |
| 149 int number_of_channels_in_frame_; | 150 int number_of_channels_in_frame_; |
| 150 | 151 |
| 151 // FIFO used to accumulates recorded data. | 152 // Dynamically allocated FIFO used when CoreAudio asks for unexpected frame |
| 152 media::AudioBlockFifo fifo_; | 153 // sizes. |
| 154 scoped_ptr<AudioBlockFifo> fifo_; |
| 155 |
| 156 // AudioBus wrapper for delievering data via AudioSourceCallback::OnData(). |
| 157 scoped_ptr<AudioBus> audio_wrapper_; |
| 153 | 158 |
| 154 // Used to defer Start() to workaround http://crbug.com/160920. | 159 // Used to defer Start() to workaround http://crbug.com/160920. |
| 155 base::CancelableClosure deferred_start_cb_; | 160 base::CancelableClosure deferred_start_cb_; |
| 156 | 161 |
| 157 DISALLOW_COPY_AND_ASSIGN(AUAudioInputStream); | 162 DISALLOW_COPY_AND_ASSIGN(AUAudioInputStream); |
| 158 }; | 163 }; |
| 159 | 164 |
| 160 } // namespace media | 165 } // namespace media |
| 161 | 166 |
| 162 #endif // MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_ | 167 #endif // MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_ |
| OLD | NEW |