/* * Copyright (C) 2011 The Android Open Source Project * Copyright (c) 2012, Code Aurora Forum. All rights reserved. * * 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_HAL_INTERFACE_H #define ANDROID_AUDIO_HAL_INTERFACE_H #include #include #include #include #include #include #include #include #include __BEGIN_DECLS /** * The id of this module */ #define AUDIO_HARDWARE_MODULE_ID "audio" /** * Name of the audio devices to open */ #define AUDIO_HARDWARE_INTERFACE "audio_hw_if" /* Use version 0.1 to be compatible with first generation of audio hw module with version_major * hardcoded to 1. No audio module API change. */ #define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1) #define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1 /* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0 * will be considered of first generation API. */ #define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0) #define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) #define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_1_0 /** * List of known audio HAL modules. This is the base name of the audio HAL * library composed of the "audio." prefix, one of the base names below and * a suffix specific to the device. * e.g: audio.primary.goldfish.so or audio.a2dp.default.so */ #define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary" #define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp" #define AUDIO_HARDWARE_MODULE_ID_USB "usb" /**************************************/ /** * standard audio parameters that the HAL may need to handle */ /** * audio device parameters */ /* BT SCO Noise Reduction + Echo Cancellation parameters */ #define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec" #define AUDIO_PARAMETER_VALUE_ON "on" #define AUDIO_PARAMETER_VALUE_OFF "off" /* TTY mode selection */ #define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode" #define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off" #define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco" #define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco" #define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full" /* A2DP sink address set by framework */ #define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address" /* Screen state */ #define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state" /** * audio stream parameters */ #define AUDIO_PARAMETER_STREAM_ROUTING "routing" // audio_devices_t #define AUDIO_PARAMETER_STREAM_FORMAT "format" // audio_format_t #define AUDIO_PARAMETER_STREAM_CHANNELS "channels" // audio_channel_mask_t #define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" // size_t #define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" // audio_source_t #define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" // uint32_t /* Query supported formats. The response is a '|' separated list of strings from * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */ #define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats" /* Query supported channel masks. The response is a '|' separated list of strings from * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */ #define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels" /* Query supported sampling rates. The response is a '|' separated list of integer values e.g: * "sup_sampling_rates=44100|48000" */ #define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates" /* Query handle fm parameter*/ #define AUDIO_PARAMETER_KEY_HANDLE_FM "handle_fm" /* Query voip flag */ #define AUDIO_PARAMETER_KEY_VOIP_CHECK "voip_flag" /* Query Fluence type */ #define AUDIO_PARAMETER_KEY_FLUENCE_TYPE "fluence" /* Query if surround sound recording is supported */ #define AUDIO_PARAMETER_KEY_SSR "ssr" /**************************************/ /* common audio stream configuration parameters */ struct audio_config { uint32_t sample_rate; audio_channel_mask_t channel_mask; audio_format_t format; }; typedef struct audio_config audio_config_t; /* common audio stream parameters and operations */ struct audio_stream { /** * Return the sampling rate in Hz - eg. 44100. */ uint32_t (*get_sample_rate)(const struct audio_stream *stream); /* currently unused - use set_parameters with key * AUDIO_PARAMETER_STREAM_SAMPLING_RATE */ int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate); /** * Return size of input/output buffer in bytes for this stream - eg. 4800. * It should be a multiple of the frame size. See also get_input_buffer_size. */ size_t (*get_buffer_size)(const struct audio_stream *stream); /** * Return the channel mask - * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO */ audio_channel_mask_t (*get_channels)(const struct audio_stream *stream); /** * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT */ audio_format_t (*get_format)(const struct audio_stream *stream); /* currently unused - use set_parameters with key * AUDIO_PARAMETER_STREAM_FORMAT */ int (*set_format)(struct audio_stream *stream, audio_format_t format); /** * Put the audio hardware input/output into standby mode. * Driver should exit from standby mode at the next I/O operation. * Returns 0 on success and <0 on failure. */ int (*standby)(struct audio_stream *stream); /** dump the state of the audio input/output device */ int (*dump)(const struct audio_stream *stream, int fd); /** Return the set of device(s) which this stream is connected to */ audio_devices_t (*get_device)(const struct audio_stream *stream); /** * Currently unused - set_device() corresponds to set_parameters() with key * AUDIO_PARAMETER_STREAM_ROUTING for both input and output. * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by * input streams only. */ int (*set_device)(struct audio_stream *stream, audio_devices_t device); /** * set/get audio stream parameters. The function accepts a list of * parameter 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 -ENOSYS. * * The audio flinger will put the stream in standby and then change the * parameter value. */ int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs); /* * Returns a pointer to a heap allocated string. The caller is responsible * for freeing the memory for it using free(). */ char * (*get_parameters)(const struct audio_stream *stream, const char *keys); int (*add_audio_effect)(const struct audio_stream *stream, effect_handle_t effect); int (*remove_audio_effect)(const struct audio_stream *stream, effect_handle_t effect); }; typedef struct audio_stream audio_stream_t; /** * audio_stream_out is the abstraction interface for the audio output hardware. * * It provides information about various properties of the audio output * hardware driver. */ struct audio_stream_out { struct audio_stream common; /** * Return the audio hardware driver estimated latency in milliseconds. */ uint32_t (*get_latency)(const struct audio_stream_out *stream); /** * 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. */ int (*set_volume)(struct audio_stream_out *stream, float left, float right); /** * Write audio buffer to driver. Returns number of bytes written, or a * negative status_t. If at least one frame was written successfully prior to the error, * it is suggested that the driver return that successful (short) byte count * and then return an error in the subsequent call. */ ssize_t (*write)(struct audio_stream_out *stream, const void* buffer, size_t bytes); /* return the number of audio frames written by the audio dsp to DAC since * the output has exited standby */ int (*get_render_position)(const struct audio_stream_out *stream, uint32_t *dsp_frames); #ifdef QCOM_HARDWARE /** * start audio data rendering */ int (*start)(struct audio_stream_out *stream); /** * pause audio rendering */ int (*pause)(struct audio_stream_out *stream); /** * flush audio data with driver */ int (*flush)(struct audio_stream_out *stream); /** * stop audio data rendering */ int (*stop)(struct audio_stream_out *stream); #endif /** * get the local time at which the next write to the audio driver will be presented. * The units are microseconds, where the epoch is decided by the local audio HAL. */ int (*get_next_write_timestamp)(const struct audio_stream_out *stream, int64_t *timestamp); #ifdef QCOM_HARDWARE /** * return the current timestamp after quering to the driver */ int (*get_time_stamp)(const struct audio_stream_out *stream, uint64_t *time_stamp); /** * EOS notification from HAL to Player */ int (*set_observer)(const struct audio_stream_out *stream, void *observer); #endif }; typedef struct audio_stream_out audio_stream_out_t; #ifdef QCOM_HARDWARE /** * audio_broadcast_stream is the abstraction interface for the * audio output hardware. * * It provides information about various properties of the audio output * hardware driver. */ struct audio_broadcast_stream { struct audio_stream common; /** * return the audio hardware driver latency in milli seconds. */ uint32_t (*get_latency)(const struct audio_broadcast_stream *stream); /** * 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. */ int (*set_volume)(struct audio_broadcast_stream *stream, float left, float right); int (*mute)(struct audio_broadcast_stream *stream, bool mute); int (*start)(struct audio_broadcast_stream *stream, int64_t absTimeToStart); /** * write audio buffer to driver. Returns number of bytes written */ ssize_t (*write)(struct audio_broadcast_stream *stream, const void* buffer, size_t bytes, int64_t timestamp, int audioType); }; typedef struct audio_broadcast_stream audio_broadcast_stream_t; #endif struct audio_stream_in { struct audio_stream common; /** set the input gain for the audio driver. This method is for * for future use */ int (*set_gain)(struct audio_stream_in *stream, float gain); /** Read audio buffer in from audio driver. Returns number of bytes read, or a * negative status_t. If at least one frame was read prior to the error, * read should return that byte count and then return an error in the subsequent call. */ ssize_t (*read)(struct audio_stream_in *stream, void* buffer, size_t bytes); /** * Return the amount 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 */ uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream); }; typedef struct audio_stream_in audio_stream_in_t; /** * return the frame size (number of bytes per sample). */ static inline size_t audio_stream_frame_size(struct audio_stream *s) { size_t chan_samp_sz; uint32_t chan_mask = s->get_channels(s); int format = s->get_format(s); #ifdef QCOM_HARDWARE if (audio_is_input_channel(chan_mask)) { chan_mask &= (AUDIO_CHANNEL_IN_STEREO | \ AUDIO_CHANNEL_IN_MONO ); } if(!strcmp(s->get_parameters(s, "voip_flag"),"voip_flag=1")) { if(format != AUDIO_FORMAT_PCM_8_BIT) return popcount(chan_mask) * sizeof(int16_t); else return popcount(chan_mask) * sizeof(int8_t); } if (audio_is_input_channel(chan_mask)) { chan_mask &= (AUDIO_CHANNEL_IN_STEREO | \ AUDIO_CHANNEL_IN_MONO | \ AUDIO_CHANNEL_IN_5POINT1); } #endif switch (format) { #ifdef QCOM_HARDWARE case AUDIO_FORMAT_AMR_NB: chan_samp_sz = 32; break; case AUDIO_FORMAT_EVRC: chan_samp_sz = 23; break; case AUDIO_FORMAT_QCELP: chan_samp_sz = 35; break; #endif case AUDIO_FORMAT_PCM_16_BIT: chan_samp_sz = sizeof(int16_t); break; case AUDIO_FORMAT_PCM_8_BIT: default: chan_samp_sz = sizeof(int8_t); break; } return popcount(chan_mask) * chan_samp_sz; } /**********************************************************************/ /** * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM * and the fields of this data structure must begin with hw_module_t * followed by module specific information. */ struct audio_module { struct hw_module_t common; }; struct audio_hw_device { struct hw_device_t common; /** * used by audio flinger to enumerate what devices are supported by * each audio_hw_device implementation. * * Return value is a bitmask of 1 or more values of audio_devices_t */ uint32_t (*get_supported_devices)(const struct audio_hw_device *dev); /** * check to see if the audio hardware interface has been initialized. * returns 0 on success, -ENODEV on failure. */ int (*init_check)(const struct audio_hw_device *dev); /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ int (*set_voice_volume)(struct audio_hw_device *dev, float volume); /** * set the audio volume for all audio activities other than voice call. * Range between 0.0 and 1.0. If any value other than 0 is returned, * the software mixer will emulate this capability. */ int (*set_master_volume)(struct audio_hw_device *dev, float volume); #ifndef ICS_AUDIO_BLOB /** * 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. HALs which do not support * this method should may leave it set to NULL. */ int (*get_master_volume)(struct audio_hw_device *dev, float *volume); #endif #ifdef QCOM_FM_ENABLED /** set the fm audio volume. Range is between 0.0 and 1.0 */ int (*set_fm_volume)(struct audio_hw_device *dev, float volume); #endif /** * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is * playing, and AUDIO_MODE_IN_CALL when a call is in progress. */ int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode); /* mic mute */ int (*set_mic_mute)(struct audio_hw_device *dev, bool state); int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state); /* set/get global audio parameters */ int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs); /* * Returns a pointer to a heap allocated string. The caller is responsible * for freeing the memory for it using free(). */ char * (*get_parameters)(const struct audio_hw_device *dev, const char *keys); /* Returns audio input buffer size according to parameters passed or * 0 if one of the parameters is not supported. * See also get_buffer_size which is for a particular stream. */ size_t (*get_input_buffer_size)(const struct audio_hw_device *dev, #ifndef ICS_AUDIO_BLOB const struct audio_config *config); #else uint32_t sample_rate, int format, int channel_count); #endif /** This method creates and opens the audio hardware output stream */ #ifndef ICS_AUDIO_BLOB int (*open_output_stream)(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, audio_output_flags_t flags, struct audio_config *config, struct audio_stream_out **stream_out); #else int (*open_output_stream)(struct audio_hw_device *dev, uint32_t devices, int *format, uint32_t *channels, uint32_t *sample_rate, struct audio_stream_out **out); #endif #ifdef QCOM_ICS_LPA_COMPAT /** This method creates and opens the audio hardware output session */ int (*open_output_session)(struct audio_hw_device *dev, uint32_t devices, int *format, int sessionId, struct audio_stream_out **out); #endif void (*close_output_stream)(struct audio_hw_device *dev, struct audio_stream_out* stream_out); #ifdef QCOM_HARDWARE /** This method creates and opens the audio hardware output * for broadcast stream */ int (*open_broadcast_stream)(struct audio_hw_device *dev, uint32_t devices, int format, uint32_t channels, uint32_t sample_rate, uint32_t audio_source, struct audio_broadcast_stream **out); void (*close_broadcast_stream)(struct audio_hw_device *dev, struct audio_broadcast_stream *out); #endif /** This method creates and opens the audio hardware input stream */ #ifndef ICS_AUDIO_BLOB int (*open_input_stream)(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, struct audio_config *config, struct audio_stream_in **stream_in); #else int (*open_input_stream)(struct audio_hw_device *dev, uint32_t devices, int *format, uint32_t *channels, uint32_t *sample_rate, audio_in_acoustics_t acoustics, struct audio_stream_in **stream_in); #endif void (*close_input_stream)(struct audio_hw_device *dev, struct audio_stream_in *stream_in); /** This method dumps the state of the audio hardware */ int (*dump)(const struct audio_hw_device *dev, int fd); }; typedef struct audio_hw_device audio_hw_device_t; /** convenience API for opening and closing a supported device */ static inline int audio_hw_device_open(const struct hw_module_t* module, struct audio_hw_device** device) { return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, (struct hw_device_t**)device); } static inline int audio_hw_device_close(struct audio_hw_device* device) { return device->common.close(&device->common); } #ifdef QCOM_HARDWARE #ifdef __cplusplus /** *Observer class to post the Events from HAL to Flinger */ class AudioEventObserver { public: virtual ~AudioEventObserver() {} virtual void postEOS(int64_t delayUs) = 0; }; #endif #endif __END_DECLS #endif // ANDROID_AUDIO_INTERFACE_H