diff options
Diffstat (limited to 'include/hardware/audio.h')
| -rw-r--r-- | include/hardware/audio.h | 210 |
1 files changed, 208 insertions, 2 deletions
diff --git a/include/hardware/audio.h b/include/hardware/audio.h index 3a0962e..beac717 100644 --- a/include/hardware/audio.h +++ b/include/hardware/audio.h @@ -1,5 +1,6 @@ /* * 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. @@ -19,6 +20,7 @@ #define ANDROID_AUDIO_HAL_INTERFACE_H #include <stdint.h> +#include <string.h> #include <strings.h> #include <sys/cdefs.h> #include <sys/types.h> @@ -54,7 +56,11 @@ __BEGIN_DECLS #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_2_0 HARDWARE_DEVICE_API_VERSION(2, 0) +#ifndef ICS_AUDIO_BLOB #define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_2_0 +#else +#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_1_0 +#endif /** * List of known audio HAL modules. This is the base name of the audio HAL @@ -117,7 +123,20 @@ __BEGIN_DECLS * "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" + +/* Query if a2dp is supported */ +#define AUDIO_PARAMETER_KEY_HANDLE_A2DP_DEVICE "isA2dpDeviceSupported" /**************************************/ /* common audio stream configuration parameters */ @@ -128,6 +147,9 @@ struct audio_config { }; typedef struct audio_config audio_config_t; +#ifdef QCOM_HARDWARE +typedef struct buf_info; +#endif /* common audio stream parameters and operations */ struct audio_stream { @@ -252,16 +274,101 @@ struct audio_stream_out { 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); + /** + * Get the physical address of the buffer allocated in the + * driver + */ + int (*get_buffer_info) (const struct audio_stream_out *stream, + struct buf_info **buf); + /** + * Check if next buffer is available. Waits until next buffer is + * available + */ + int (*is_buffer_available) (const struct audio_stream_out *stream, + int *isAvail); +#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; @@ -296,8 +403,39 @@ typedef struct audio_stream_in audio_stream_in_t; static inline size_t audio_stream_frame_size(const struct audio_stream *s) { size_t chan_samp_sz; + uint32_t chan_mask = s->get_channels(s); + int format = s->get_format(s); - switch (s->get_format(s)) { +#ifdef QCOM_HARDWARE + if (!s) + return 0; + + if (audio_is_input_channel(chan_mask)) { + chan_mask &= (AUDIO_CHANNEL_IN_STEREO | \ + AUDIO_CHANNEL_IN_MONO | \ + AUDIO_CHANNEL_IN_5POINT1); + } + + if(!strncmp(s->get_parameters(s, "voip_flag"),"voip_flag=1",sizeof("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); + } +#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; @@ -307,7 +445,7 @@ static inline size_t audio_stream_frame_size(const struct audio_stream *s) break; } - return popcount(s->get_channels(s)) * chan_samp_sz; + return popcount(chan_mask) * chan_samp_sz; } @@ -355,6 +493,7 @@ struct audio_hw_device { */ 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 @@ -363,6 +502,12 @@ struct audio_hw_device { * this method 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 @@ -390,25 +535,65 @@ struct audio_hw_device { * 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); @@ -416,6 +601,7 @@ struct audio_hw_device { /** This method dumps the state of the audio hardware */ int (*dump)(const struct audio_hw_device *dev, int fd); +#ifndef ICS_AUDIO_BLOB /** * set the audio mute status for all audio activities. If any value other * than 0 is returned, the software mixer will emulate this capability. @@ -430,6 +616,7 @@ struct audio_hw_device { * method may leave it set to NULL. */ int (*get_master_mute)(struct audio_hw_device *dev, bool *mute); +#endif }; typedef struct audio_hw_device audio_hw_device_t; @@ -447,7 +634,26 @@ static inline int audio_hw_device_close(struct audio_hw_device* device) return device->common.close(&device->common); } +#ifdef QCOM_HARDWARE +/** Structure to save buffer information for applying effects for + * LPA buffers */ +struct buf_info { + int bufsize; + int nBufs; + int **buffers; +}; +#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 |