hal/audio_hw.h

/*
 * Copyright (C) 2013 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 QCOM_AUDIO_HW_H
#define QCOM_AUDIO_HW_H

#include <cutils/list.h>
#include <hardware/audio.h>

#include <tinyalsa/asoundlib.h>
#include <tinycompress/tinycompress.h>

#include <audio_route/audio_route.h>

/* Flags used to initialize acdb_settings variable that goes to ACDB library */
#define DMIC_FLAG       0x00000002
#define TTY_MODE_OFF    0x00000010
#define TTY_MODE_FULL   0x00000020
#define TTY_MODE_VCO    0x00000040
#define TTY_MODE_HCO    0x00000080
#define TTY_MODE_CLEAR  0xFFFFFF0F

#define ACDB_DEV_TYPE_OUT 1
#define ACDB_DEV_TYPE_IN 2

#define MAX_SUPPORTED_CHANNEL_MASKS 2

typedef int snd_device_t;

/* These are the supported use cases by the hardware.
 * Each usecase is mapped to a specific PCM device.
 * Refer to pcm_device_table[].
 */
typedef enum {
    USECASE_INVALID = -1,
    /* Playback usecases */
    USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
    USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
    USECASE_AUDIO_PLAYBACK_MULTI_CH,
    USECASE_AUDIO_PLAYBACK_OFFLOAD,

    /* Capture usecases */
    USECASE_AUDIO_RECORD,
    USECASE_AUDIO_RECORD_LOW_LATENCY,

    USECASE_VOICE_CALL,
    AUDIO_USECASE_MAX
} audio_usecase_t;

#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))

/*
 * tinyAlsa library interprets period size as number of frames
 * one frame = channel_count * sizeof (pcm sample)
 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
 * We should take care of returning proper size when AudioFlinger queries for
 * the buffer size of an input/output stream
 */

enum {
    OFFLOAD_CMD_EXIT,               /* exit compress offload thread loop*/
    OFFLOAD_CMD_DRAIN,              /* send a full drain request to DSP */
    OFFLOAD_CMD_PARTIAL_DRAIN,      /* send a partial drain request to DSP */
    OFFLOAD_CMD_WAIT_FOR_BUFFER,    /* wait for buffer released by DSP */
};

enum {
    OFFLOAD_STATE_IDLE,
    OFFLOAD_STATE_PLAYING,
    OFFLOAD_STATE_PAUSED,
};

struct offload_cmd {
    struct listnode node;
    int cmd;
    int data[];
};

struct stream_out {
    struct audio_stream_out stream;
    pthread_mutex_t lock; /* see note below on mutex acquisition order */
    pthread_cond_t  cond;
    struct pcm_config config;
    struct compr_config compr_config;
    struct pcm *pcm;
    struct compress *compr;
    int standby;
    int pcm_device_id;
    unsigned int sample_rate;
    audio_channel_mask_t channel_mask;
    audio_format_t format;
    audio_devices_t devices;
    audio_output_flags_t flags;
    audio_usecase_t usecase;
    /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
    audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
    bool muted;

    int non_blocking;
    int playback_started;
    int offload_state;
    pthread_cond_t offload_cond;
    pthread_t offload_thread;
    struct listnode offload_cmd_list;
    bool offload_thread_blocked;

    stream_callback_t offload_callback;
    void *offload_cookie;

    struct audio_device *dev;
};

struct stream_in {
    struct audio_stream_in stream;
    pthread_mutex_t lock; /* see note below on mutex acquisition order */
    struct pcm_config config;
    struct pcm *pcm;
    int standby;
    int source;
    int pcm_device_id;
    int device;
    audio_channel_mask_t channel_mask;
    audio_usecase_t usecase;
    bool enable_aec;

    struct audio_device *dev;
};

typedef enum {
    PCM_PLAYBACK,
    PCM_CAPTURE,
    VOICE_CALL
} usecase_type_t;

union stream_ptr {
    struct stream_in *in;
    struct stream_out *out;
};

struct audio_usecase {
    struct listnode list;
    audio_usecase_t id;
    usecase_type_t  type;
    audio_devices_t devices;
    snd_device_t out_snd_device;
    snd_device_t in_snd_device;
    union stream_ptr stream;
};

struct audio_device {
    struct audio_hw_device device;
    pthread_mutex_t lock; /* see note below on mutex acquisition order */
    struct mixer *mixer;
    audio_mode_t mode;
    audio_devices_t out_device;
    struct stream_in *active_input;
    struct stream_out *primary_output;
    int in_call;
    float voice_volume;
    bool mic_mute;
    int tty_mode;
    bool bluetooth_nrec;
    bool screen_off;
    struct pcm *voice_call_rx;
    struct pcm *voice_call_tx;
    int *snd_dev_ref_cnt;
    struct listnode usecase_list;
    struct audio_route *audio_route;
    int acdb_settings;
    bool speaker_lr_swap;

    void *platform;
};

/*
 * NOTE: when multiple mutexes have to be acquired, always take the
 * stream_in or stream_out mutex first, followed by the audio_device mutex.
 */

#endif // QCOM_AUDIO_HW_H