hal: Add platform specific code for 8909

/*

 *

 * 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_PLATFORM_H

#define QCOM_AUDIO_PLATFORM_H

enum {

    FLUENCE_NONE,

    FLUENCE_DUAL_MIC = 0x1,

    FLUENCE_QUAD_MIC = 0x2,

};

enum {

    FLUENCE_ENDFIRE = 0x1,

    FLUENCE_BROADSIDE = 0x2,

};

/*

 * Below are the devices for which is back end is same, SLIMBUS_0_RX.

 * All these devices are handled by the internal HW codec. We can

 * enable any one of these devices at any time. An exception here is

 * 44.1k headphone which uses different backend. This is filtered

 * as different hal internal device in the code but remains same

 * as standard android device AUDIO_DEVICE_OUT_WIRED_HEADPHONE

 * for other layers.

 */

#define AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND \

    (AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \

     AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE|\

     AUDIO_DEVICE_OUT_LINE)

/* Sound devices specific to the platform

 * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound

 * devices to enable corresponding mixer paths

 */

enum {

    SND_DEVICE_NONE = 0,

    /* Playback devices */

    SND_DEVICE_MIN,

    SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN,

    SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN,

    SND_DEVICE_OUT_SPEAKER,

    SND_DEVICE_OUT_SPEAKER_REVERSE,

    SND_DEVICE_OUT_LINE,

    SND_DEVICE_OUT_HEADPHONES,

    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,

    SND_DEVICE_OUT_SPEAKER_AND_LINE,

    SND_DEVICE_OUT_VOICE_HANDSET,

    SND_DEVICE_OUT_VOICE_HAC_HANDSET,

    SND_DEVICE_OUT_VOICE_SPEAKER,

    SND_DEVICE_OUT_VOICE_HEADPHONES,

    SND_DEVICE_OUT_VOICE_LINE,

    SND_DEVICE_OUT_HDMI,

    SND_DEVICE_OUT_SPEAKER_AND_HDMI,

    SND_DEVICE_OUT_BT_SCO,

    SND_DEVICE_OUT_BT_SCO_WB,

    SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,

    SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,

    SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,

    SND_DEVICE_OUT_VOICE_TX,

    SND_DEVICE_OUT_AFE_PROXY,

    SND_DEVICE_OUT_USB_HEADSET,

    SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,

    SND_DEVICE_OUT_ANC_HEADSET,

    SND_DEVICE_OUT_ANC_FB_HEADSET,

    SND_DEVICE_OUT_VOICE_ANC_HEADSET,

    SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET,

    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,

    SND_DEVICE_OUT_ANC_HANDSET,

    SND_DEVICE_OUT_SPEAKER_PROTECTED,

    SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,

    SND_DEVICE_OUT_END,

    /*

     * Note: IN_BEGIN should be same as OUT_END because total number of devices

     * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices.

     */

    /* Capture devices */

    SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END,

    SND_DEVICE_IN_HANDSET_MIC  = SND_DEVICE_IN_BEGIN,

    SND_DEVICE_IN_HANDSET_MIC_EXTERNAL,

    SND_DEVICE_IN_HANDSET_MIC_AEC,

    SND_DEVICE_IN_HANDSET_MIC_NS,

    SND_DEVICE_IN_HANDSET_MIC_AEC_NS,

    SND_DEVICE_IN_HANDSET_DMIC,

    SND_DEVICE_IN_HANDSET_DMIC_AEC,

    SND_DEVICE_IN_HANDSET_DMIC_NS,

    SND_DEVICE_IN_HANDSET_DMIC_AEC_NS,

    SND_DEVICE_IN_SPEAKER_MIC,

    SND_DEVICE_IN_SPEAKER_MIC_AEC,

    SND_DEVICE_IN_SPEAKER_MIC_NS,

    SND_DEVICE_IN_SPEAKER_MIC_AEC_NS,

    SND_DEVICE_IN_SPEAKER_DMIC,

    SND_DEVICE_IN_SPEAKER_DMIC_AEC,

    SND_DEVICE_IN_SPEAKER_DMIC_NS,

    SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS,

    SND_DEVICE_IN_HEADSET_MIC,

    SND_DEVICE_IN_HEADSET_MIC_FLUENCE,

    SND_DEVICE_IN_VOICE_SPEAKER_MIC,

    SND_DEVICE_IN_VOICE_HEADSET_MIC,

    SND_DEVICE_IN_HDMI_MIC,

    SND_DEVICE_IN_BT_SCO_MIC,

    SND_DEVICE_IN_BT_SCO_MIC_NREC,

    SND_DEVICE_IN_BT_SCO_MIC_WB,

    SND_DEVICE_IN_BT_SCO_MIC_WB_NREC,

    SND_DEVICE_IN_CAMCORDER_MIC,

    SND_DEVICE_IN_VOICE_DMIC,

    SND_DEVICE_IN_VOICE_SPEAKER_DMIC,

    SND_DEVICE_IN_VOICE_SPEAKER_QMIC,

    SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,

    SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,

    SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,

    SND_DEVICE_IN_VOICE_REC_MIC,

    SND_DEVICE_IN_VOICE_REC_MIC_NS,

    SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,

    SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,

    SND_DEVICE_IN_VOICE_RX,

    SND_DEVICE_IN_USB_HEADSET_MIC,

    SND_DEVICE_IN_CAPTURE_FM,

    SND_DEVICE_IN_AANC_HANDSET_MIC,

    SND_DEVICE_IN_QUAD_MIC,

    SND_DEVICE_IN_HANDSET_STEREO_DMIC,

    SND_DEVICE_IN_SPEAKER_STEREO_DMIC,

    SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,

    SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,

    SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,

    SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,

    SND_DEVICE_IN_SPEAKER_DMIC_NS_BROADSIDE,

    SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS_BROADSIDE,

    SND_DEVICE_IN_VOICE_FLUENCE_DMIC_AANC,

    SND_DEVICE_IN_HANDSET_QMIC,

    SND_DEVICE_IN_SPEAKER_QMIC_AEC,

    SND_DEVICE_IN_SPEAKER_QMIC_NS,

    SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS,

    SND_DEVICE_IN_END,

    SND_DEVICE_MAX = SND_DEVICE_IN_END,

};

#define DEFAULT_OUTPUT_SAMPLING_RATE 48000

#define ALL_SESSION_VSID                0xFFFFFFFF

#define DEFAULT_MUTE_RAMP_DURATION_MS   20

#define DEFAULT_VOLUME_RAMP_DURATION_MS 20

#define MIXER_PATH_MAX_LENGTH 100

#define MAX_VOL_INDEX 5

#define MIN_VOL_INDEX 0

#define percent_to_index(val, min, max) \

            ((val) * ((max) - (min)) * 0.01 + (min) + .5)

/*

 * 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

 */

#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920

#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2

#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240

#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2

#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000

#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240

#define LOW_LATENCY_CAPTURE_USE_CASE 1

#define HDMI_MULTI_PERIOD_SIZE  336

#define HDMI_MULTI_PERIOD_COUNT 8

#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6

#define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)

#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20

#define AUDIO_CAPTURE_PERIOD_COUNT 2

#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000

#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240

#define LOW_LATENCY_CAPTURE_USE_CASE 1

#define DEVICE_NAME_MAX_SIZE 128

#define HW_INFO_ARRAY_MAX_SIZE 32

#define DEEP_BUFFER_PCM_DEVICE 0

#define AUDIO_RECORD_PCM_DEVICE 0

#define MULTIMEDIA2_PCM_DEVICE 1

#define MULTIMEDIA3_PCM_DEVICE 4

#define FM_PLAYBACK_PCM_DEVICE 5

#define FM_CAPTURE_PCM_DEVICE  6

#define HFP_PCM_RX 5

#define HFP_SCO_RX 17

#define HFP_ASM_RX_TX 18

#define INCALL_MUSIC_UPLINK_PCM_DEVICE 1

#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16

#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5

#define SPKR_PROT_CALIB_TX_PCM_DEVICE 26

#define PLAYBACK_OFFLOAD_DEVICE 9

#define PLAYBACK_OFFLOAD_DEVICE2 24

/* Define macro for Internal FM volume mixer */

#define FM_RX_VOLUME "Internal FM RX Volume"

#define LOWLATENCY_PCM_DEVICE 12

#define EC_REF_RX "I2S_RX"

#define VOICE_CALL_PCM_DEVICE 2

#define VOICE2_CALL_PCM_DEVICE 13

#define VOLTE_CALL_PCM_DEVICE 15

#define QCHAT_CALL_PCM_DEVICE 26

#define QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC 28

#define VOWLAN_CALL_PCM_DEVICE 16

#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7

#define AFE_PROXY_RECORD_PCM_DEVICE 8

#define LIB_CSD_CLIENT "libcsd-client.so"

/* CSD-CLIENT related functions */

typedef int (*init_t)();

typedef int (*deinit_t)();

typedef int (*disable_device_t)();

typedef int (*enable_device_config_t)(int, int);

typedef int (*enable_device_t)(int, int, uint32_t);

typedef int (*volume_t)(uint32_t, int, uint16_t);

typedef int (*mic_mute_t)(uint32_t, int, uint16_t);

typedef int (*slow_talk_t)(uint32_t, uint8_t);

typedef int (*start_voice_t)(uint32_t);

typedef int (*stop_voice_t)(uint32_t);

typedef int (*start_playback_t)(uint32_t);

typedef int (*stop_playback_t)(uint32_t);

typedef int (*start_record_t)(uint32_t, int);

typedef int (*stop_record_t)(uint32_t);

/* CSD Client structure */

struct csd_data {

    void *csd_client;

    init_t init;

    deinit_t deinit;

    disable_device_t disable_device;

    enable_device_config_t enable_device_config;

    enable_device_t enable_device;

    volume_t volume;

    mic_mute_t mic_mute;

    slow_talk_t slow_talk;

    start_voice_t start_voice;

    stop_voice_t stop_voice;

    start_playback_t start_playback;

    stop_playback_t stop_playback;

    start_record_t start_record;

    stop_record_t stop_record;

};

#define PLATFORM_INFO_XML_PATH          "/system/etc/audio_platform_info.xml"

#define PLATFORM_INFO_XML_BASE_STRING   "/system/etc/audio_platform_info"

#endif // QCOM_AUDIO_PLATFORM_H