#include <utils/Errors.h>
#include <utils/Vector.h>
#include <utils/String16.h>
+#include <utils/String8.h>
#include <media/IAudioFlinger.h>
#include "media/AudioSystem.h"
virtual size_t bufferSize() const = 0;
/**
- * return number of output audio channels.
- * Acceptable values are 1 (mono) or 2 (stereo)
+ * returns the output channel nask
*/
- virtual int channelCount() const = 0;
+ virtual uint32_t channels() const = 0;
/**
* return audio format in 8bit or 16bit PCM format -
/**
* return the frame size (number of bytes per sample).
*/
- uint32_t frameSize() const { return channelCount()*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
+ uint32_t frameSize() const { return AudioSystem::popCount(channels())*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
/**
* return the audio hardware driver latency in milli seconds.
* This method might produce multiple PCM outputs or hardware accelerated
* codecs, such as MP3 or AAC.
*/
- virtual status_t setVolume(float volume) = 0;
+ virtual status_t setVolume(float left, float right) = 0;
/** write audio buffer to driver. Returns number of bytes written */
virtual ssize_t write(const void* buffer, size_t bytes) = 0;
/** dump the state of the audio output device */
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
+
+ // set/get audio output parameters. The function accepts a list of parameters
+ // key value pairs in the form: key1=value1;key2=value2;...
+ // Some keys are reserved for standard parameters (See AudioParameter class).
+ // If the implementation does not accept a parameter change while the output is
+ // active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
+ // The audio flinger will put the output in standby and then change the parameter value.
+ virtual status_t setParameters(const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(const String8& keys) = 0;
};
/**
public:
virtual ~AudioStreamIn() = 0;
+ /** return audio sampling rate in hz - eg. 44100 */
+ virtual uint32_t sampleRate() const = 0;
+
/** return the input buffer size allowed by audio driver */
virtual size_t bufferSize() const = 0;
- /** return the number of audio input channels */
- virtual int channelCount() const = 0;
+ /** return input channel mask */
+ virtual uint32_t channels() const = 0;
/**
* return audio format in 8bit or 16bit PCM format -
/**
* return the frame size (number of bytes per sample).
*/
- uint32_t frameSize() const { return channelCount()*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
+ uint32_t frameSize() const { return AudioSystem::popCount(channels())*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
/** set the input gain for the audio driver. This method is for
* for future use */
*/
virtual status_t standby() = 0;
+ // set/get audio input parameters. The function accepts a list of parameters
+ // key value pairs in the form: key1=value1;key2=value2;...
+ // Some keys are reserved for standard parameters (See AudioParameter class).
+ // If the implementation does not accept a parameter change while the output is
+ // active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
+ // The audio flinger will put the input in standby and then change the parameter value.
+ virtual status_t setParameters(const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(const String8& keys) = 0;
};
/**
virtual status_t setMasterVolume(float volume) = 0;
/**
- * Audio routing methods. Routes defined in include/hardware_legacy/AudioSystem.h.
- * Audio routes can be (ROUTE_EARPIECE | ROUTE_SPEAKER | ROUTE_BLUETOOTH
- * | ROUTE_HEADSET)
- *
- * setRouting sets the routes for a mode. This is called at startup. It is
- * also called when a new device is connected, such as a wired headset is
- * plugged in or a Bluetooth headset is paired.
- */
- virtual status_t setRouting(int mode, uint32_t routes) = 0;
-
- virtual status_t getRouting(int mode, uint32_t* routes) = 0;
-
- /**
* setMode is called when the audio mode changes. NORMAL mode is for
* standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
* when a call is in progress.
*/
virtual status_t setMode(int mode) = 0;
- virtual status_t getMode(int* mode) = 0;
// mic mute
virtual status_t setMicMute(bool state) = 0;
virtual status_t getMicMute(bool* state) = 0;
- // Temporary interface, do not use
- // TODO: Replace with a more generic key:value get/set mechanism
- virtual status_t setParameter(const char* key, const char* value) = 0;
+ // set/get global audio parameters
+ virtual status_t setParameters(const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(const String8& keys) = 0;
// Returns audio input buffer size according to parameters passed or 0 if one of the
// parameters is not supported
virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
-
+
/** This method creates and opens the audio hardware output stream */
virtual AudioStreamOut* openOutputStream(
- int format=0,
- int channelCount=0,
- uint32_t sampleRate=0,
+ uint32_t devices,
+ int *format=0,
+ uint32_t *channels=0,
+ uint32_t *sampleRate=0,
status_t *status=0) = 0;
-
+ virtual void closeOutputStream(AudioStreamOut* out) = 0;
/** This method creates and opens the audio hardware input stream */
virtual AudioStreamIn* openInputStream(
- int inputSource,
- int format,
- int channelCount,
- uint32_t sampleRate,
+ uint32_t devices,
+ int *format,
+ uint32_t *channels,
+ uint32_t *sampleRate,
status_t *status,
AudioSystem::audio_in_acoustics acoustics) = 0;
+ virtual void closeInputStream(AudioStreamIn* in) = 0;
/**This method dumps the state of the audio hardware */
virtual status_t dumpState(int fd, const Vector<String16>& args) = 0;
static AudioHardwareInterface* create();
protected:
- /**
- * doRouting actually initiates the routing. A call to setRouting
- * or setMode may result in a routing change. The generic logic calls
- * doRouting when required. If the device has any special requirements these
- * methods can be overriden.
- */
- virtual status_t doRouting() = 0;
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
};
--- /dev/null
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYINTERFACE_H
+#define ANDROID_AUDIOPOLICYINTERFACE_H
+
+#include <media/AudioSystem.h>
+#include <media/ToneGenerator.h>
+#include <utils/String8.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+
+// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
+// between the platform specific audio policy manager and Android generic audio policy manager.
+// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
+// This implementation makes use of the AudioPolicyClientInterface to control the activity and
+// configuration of audio input and output streams.
+//
+// The platform specific audio policy manager is in charge of the audio routing and volume control
+// policies for a given platform.
+// The main roles of this module are:
+// - keep track of current system state (removable device connections, phone state, user requests...).
+// System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
+// - process getOutput() queries received when AudioTrack objects are created: Those queries
+// return a handler on an output that has been selected, configured and opened by the audio policy manager and that
+// must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
+// When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
+// to close or reconfigure the output depending on other streams using this output and current system state.
+// - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
+// - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
+// applicable to each output as a function of platform specific settings and current output route (destination device). It
+// also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
+//
+// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
+// and is linked with libaudioflinger.so
+
+
+// Audio Policy Manager Interface
+class AudioPolicyInterface
+{
+
+public:
+ virtual ~AudioPolicyInterface() {}
+ //
+ // configuration functions
+ //
+
+ // indicate a change in device connection status
+ virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address) = 0;
+ // retreive a device connection status
+ virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+ const char *device_address) = 0;
+ // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode
+ virtual void setPhoneState(int state) = 0;
+ // indicate a change in ringer mode
+ virtual void setRingerMode(uint32_t mode, uint32_t mask) = 0;
+ // force using a specific device category for the specified usage
+ virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0;
+ // retreive current device category forced for a given usage
+ virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0;
+ // set a system property (e.g. camera sound always audible)
+ virtual void setSystemProperty(const char* property, const char* value) = 0;
+
+
+ //
+ // Audio routing query functions
+ //
+
+ // request an output appriate for playback of the supplied stream type and parameters
+ virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate = 0,
+ uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT) = 0;
+ // indicates to the audio policy manager that the output starts being used by corresponding stream.
+ virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0;
+ // indicates to the audio policy manager that the output stops being used by corresponding stream.
+ virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0;
+ // releases the output.
+ virtual void releaseOutput(audio_io_handle_t output) = 0;
+
+ // request an input appriate for record from the supplied device with supplied parameters.
+ virtual audio_io_handle_t getInput(int inputSource,
+ uint32_t samplingRate = 0,
+ uint32_t Format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0) = 0;
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input) = 0;
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input) = 0;
+ // releases the input.
+ virtual void releaseInput(audio_io_handle_t input) = 0;
+
+ //
+ // volume control functions
+ //
+
+ // initialises stream volume conversion parameters by specifying volume index range.
+ virtual void initStreamVolume(AudioSystem::stream_type stream,
+ int indexMin,
+ int indexMax) = 0;
+
+ // sets the new stream volume at a level corresponding to the supplied index
+ virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0;
+ // retreive current volume index for the specified stream
+ virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0;
+};
+
+
+// Audio Policy client Interface
+class AudioPolicyClientInterface
+{
+public:
+ virtual ~AudioPolicyClientInterface() {}
+
+ //
+ // Audio output Control functions
+ //
+
+ // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
+ // in case the audio policy manager has no specific requirements for the output being opened.
+ // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
+ // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ AudioSystem::output_flags flags) = 0;
+ // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
+ // a special mixer thread in the AudioFlinger.
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
+ // closes the output stream
+ virtual status_t closeOutput(audio_io_handle_t output) = 0;
+ // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
+ // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
+ virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+ // restores a suspended output.
+ virtual status_t restoreOutput(audio_io_handle_t output) = 0;
+
+ //
+ // Audio input Control functions
+ //
+
+ // opens an audio input
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t acoustics) = 0;
+ // closes an audio input
+ virtual status_t closeInput(audio_io_handle_t input) = 0;
+ //
+ // misc control functions
+ //
+
+ // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
+ // for each output (destination device) it is attached to.
+ virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output) = 0;
+
+ // reroute a given stream type to the specified output
+ virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output) = 0;
+
+ // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
+ virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) = 0;
+ // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
+
+ // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
+ // over a telephony device during a phone call.
+ virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0;
+ virtual status_t stopTone() = 0;
+};
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface);
+
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYINTERFACE_H
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