| /* |
| * Copyright (C) 2007 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_AUDIO_HARDWARE_INTERFACE_H |
| #define ANDROID_AUDIO_HARDWARE_INTERFACE_H |
| |
| #include <stdint.h> |
| #include <sys/types.h> |
| |
| #include <utils/Errors.h> |
| #include <utils/Vector.h> |
| #include <utils/String16.h> |
| #include <utils/String8.h> |
| |
| #include <media/IAudioFlinger.h> |
| #include <hardware_legacy/AudioSystemLegacy.h> |
| |
| #include <system/audio.h> |
| #include <hardware/audio.h> |
| |
| #include <cutils/bitops.h> |
| |
| namespace android_audio_legacy { |
| using android::Vector; |
| using android::String16; |
| using android::String8; |
| |
| // ---------------------------------------------------------------------------- |
| |
| /** |
| * AudioStreamOut is the abstraction interface for the audio output hardware. |
| * |
| * It provides information about various properties of the audio output hardware driver. |
| */ |
| class AudioStreamOut { |
| public: |
| virtual ~AudioStreamOut() = 0; |
| |
| /** return audio sampling rate in hz - eg. 44100 */ |
| virtual uint32_t sampleRate() const = 0; |
| |
| /** returns size of output buffer - eg. 4800 */ |
| virtual size_t bufferSize() const = 0; |
| |
| /** |
| * returns the output channel mask |
| */ |
| virtual uint32_t channels() const = 0; |
| |
| /** |
| * return audio format in 8bit or 16bit PCM format - |
| * eg. AudioSystem:PCM_16_BIT |
| */ |
| virtual int format() const = 0; |
| |
| /** |
| * return the frame size (number of bytes per sample). |
| */ |
| uint32_t frameSize() const { return popcount(channels())*((format()==AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); } |
| |
| /** |
| * return the audio hardware driver latency in milli seconds. |
| */ |
| virtual uint32_t latency() const = 0; |
| |
| /** |
| * Use this method in situations where audio mixing is done in the |
| * hardware. This method serves as a direct interface with hardware, |
| * allowing you to directly set the volume as apposed to via the framework. |
| * This method might produce multiple PCM outputs or hardware accelerated |
| * codecs, such as MP3 or AAC. |
| */ |
| 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; |
| |
| /** |
| * Put the audio hardware output into standby mode. Returns |
| * status based on include/utils/Errors.h |
| */ |
| virtual status_t standby() = 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; |
| |
| // return the number of audio frames written by the audio dsp to DAC since |
| // the output has exited standby |
| virtual status_t getRenderPosition(uint32_t *dspFrames) = 0; |
| |
| /** |
| * get the local time at which the next write to the audio driver will be |
| * presented |
| */ |
| virtual status_t getNextWriteTimestamp(int64_t *timestamp); |
| |
| }; |
| |
| /** |
| * AudioStreamIn is the abstraction interface for the audio input hardware. |
| * |
| * It defines the various properties of the audio hardware input driver. |
| */ |
| class AudioStreamIn { |
| 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 input channel mask */ |
| virtual uint32_t channels() const = 0; |
| |
| /** |
| * return audio format in 8bit or 16bit PCM format - |
| * eg. AudioSystem:PCM_16_BIT |
| */ |
| virtual int format() const = 0; |
| |
| /** |
| * return the frame size (number of bytes per sample). |
| */ |
| 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 setGain(float gain) = 0; |
| |
| /** read audio buffer in from audio driver */ |
| virtual ssize_t read(void* buffer, ssize_t bytes) = 0; |
| |
| /** dump the state of the audio input device */ |
| virtual status_t dump(int fd, const Vector<String16>& args) = 0; |
| |
| /** |
| * Put the audio hardware input into standby mode. Returns |
| * status based on include/utils/Errors.h |
| */ |
| 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; |
| |
| |
| // Return the number of input frames lost in the audio driver since the last call of this function. |
| // Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call. |
| // Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers. |
| // Unit: the number of input audio frames |
| virtual unsigned int getInputFramesLost() const = 0; |
| |
| virtual status_t addAudioEffect(effect_handle_t effect) = 0; |
| virtual status_t removeAudioEffect(effect_handle_t effect) = 0; |
| }; |
| |
| /** |
| * AudioHardwareInterface.h defines the interface to the audio hardware abstraction layer. |
| * |
| * The interface supports setting and getting parameters, selecting audio routing |
| * paths, and defining input and output streams. |
| * |
| * AudioFlinger initializes the audio hardware and immediately opens an output stream. |
| * You can set Audio routing to output to handset, speaker, Bluetooth, or a headset. |
| * |
| * The audio input stream is initialized when AudioFlinger is called to carry out |
| * a record operation. |
| */ |
| class AudioHardwareInterface |
| { |
| public: |
| virtual ~AudioHardwareInterface() {} |
| |
| /** |
| * check to see if the audio hardware interface has been initialized. |
| * return status based on values defined in include/utils/Errors.h |
| */ |
| virtual status_t initCheck() = 0; |
| |
| /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ |
| virtual status_t setVoiceVolume(float volume) = 0; |
| |
| /** |
| * set the audio volume for all audio activities other than voice call. |
| * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned, |
| * the software mixer will emulate this capability. |
| */ |
| virtual status_t setMasterVolume(float volume) = 0; |
| |
| /** |
| * Get the current master volume value for the HAL, if the HAL supports |
| * master volume control. AudioFlinger will query this value from the |
| * primary audio HAL when the service starts and use the value for setting |
| * the initial master volume across all HALs. |
| */ |
| virtual status_t getMasterVolume(float *volume) = 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; |
| |
| // mic mute |
| virtual status_t setMicMute(bool state) = 0; |
| virtual status_t getMicMute(bool* state) = 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( |
| 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( |
| 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: |
| |
| virtual status_t dump(int fd, const Vector<String16>& args) = 0; |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| |
| extern "C" AudioHardwareInterface* createAudioHardware(void); |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIO_HARDWARE_INTERFACE_H |