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/*
* Copyright (c) 2013, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* 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,
};
#include <hardware/audio.h>
/*
* 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
*/
#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)
/*TODO remove this once define in audio.h */
#define AUDIO_DEVICE_OUT_SPDIF 0x4000
/* 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_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_VOICE_HANDSET,
SND_DEVICE_OUT_VOICE_SPEAKER,
SND_DEVICE_OUT_VOICE_HEADPHONES,
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_AFE_PROXY,
SND_DEVICE_OUT_USB_HEADSET,
SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,
SND_DEVICE_OUT_TRANSMISSION_FM,
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_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_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_WB,
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_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_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
};
#define MIXER_CARD 0
#define SOUND_CARD 0
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION 500
#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 960
#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
/*******************************************************************************
ADTS HEADER PARSING
*******************************************************************************/
//Required for ADTS Header Parsing
#define ADTS_HEADER_SYNC_RESULT 0xfff0
#define ADTS_HEADER_SYNC_MASK 0xfff6
/*******************************************************************************
HDMI and SPDIF Device Output format control
*******************************************************************************/
#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 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 FM_PLAYBACK_PCM_DEVICE 5
#define FM_CAPTURE_PCM_DEVICE 6
#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 22
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
#define MULTI_CHANNEL_PCM_DEVICE 1
#define VOICE_CALL_PCM_DEVICE 2
//TODO: update the device number as per the dai links
#define PLAYBACK_OFFLOAD_DEVICE1 2
#define PLAYBACK_OFFLOAD_DEVICE2 3
#define PLAYBACK_OFFLOAD_DEVICE3 4
#define PLAYBACK_OFFLOAD_DEVICE4 19
#define LOWLATENCY_PCM_DEVICE 15
#define COMPRESS_CAPTURE_DEVICE 19
#ifdef PLATFORM_MSM8x26
#define VOICE_CALL_PCM_DEVICE 2
#define VOICE2_CALL_PCM_DEVICE 14
#define VOLTE_CALL_PCM_DEVICE 17
#define QCHAT_CALL_PCM_DEVICE 18
#elif PLATFORM_APQ8084
#define VOICE_CALL_PCM_DEVICE 20
#define VOICE2_CALL_PCM_DEVICE 13
#define VOLTE_CALL_PCM_DEVICE 21
#define QCHAT_CALL_PCM_DEVICE 06
#else
#define VOICE_CALL_PCM_DEVICE 2
#define VOICE2_CALL_PCM_DEVICE 13
#define VOLTE_CALL_PCM_DEVICE 14
#define QCHAT_CALL_PCM_DEVICE 20
#endif
#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_t)(int, int, uint32_t);
typedef int (*volume_t)(uint32_t, int);
typedef int (*mic_mute_t)(uint32_t, int);
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, int);
/* CSD Client structure */
struct csd_data {
void *csd_client;
init_t init;
deinit_t deinit;
disable_device_t disable_device;
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;
};
void *hw_info_init(const char *snd_card_name);
void hw_info_deinit(void *hw_info);
void hw_info_append_hw_type(void *hw_info, snd_device_t snd_device,
char *device_name);
/*******************************************************************************
USECASES AND THE CORRESPONDING DEVICE FORMATS THAT WE SUPPORT IN HAL
*******************************************************************************/
/*
In general max of 2 for pass through. Additional 1 for handling transcode
as the existence of transcode is with a PCM handle followed by transcode handle
So, a (AC3/EAC3) pass through + trancode require - 1 for pas through, 1 - pcm and
1 - transcode
*/
#define NUM_DEVICES_SUPPORT_COMPR_DATA 2+1
#define NUM_SUPPORTED_CODECS 16
#define NUM_COLUMN_FOR_INDEXING 2
#define NUM_STATES_FOR_EACH_DEVICE_FMT 3
#define DECODER_TYPE_IDX 0
#define ROUTE_FORMAT_IDX 1
#define MIN_SIZE_FOR_METADATA 64
#define NUM_OF_PERIODS 8
/*Period size to be a multiple of chanels * bitwidth,
So min period size = LCM (1,2...8) * 4*/
#define PERIOD_SIZE_COMPR 3360
#define MS11_INPUT_BUFFER_SIZE 1536
/*Max Period size which is exposed by the compr driver
The value needs to be modified when the period size is modified*/
#define PLAYBACK_MAX_PERIOD_SIZE (160 * 1024)
#define COMPR_INPUT_BUFFER_SIZE (PERIOD_SIZE_COMPR - MIN_SIZE_FOR_METADATA)
#define PCM_16_BITS_PER_SAMPLE 2
#define PCM_24_BITS_PER_SAMPLE 3
#define AC3_PERIOD_SIZE 1536 * PCM_16_BITS_PER_SAMPLE
#define TIME_PER_BUFFER 40 //Time duration in ms
#define SAMPLES_PER_CHANNEL 32*1024 //1536*2 /*TODO:correct it
#define MAX_INPUT_CHANNELS_SUPPORTED 8
#define FACTOR_FOR_BUFFERING 2
#define STEREO_CHANNELS 2
#define MAX_OUTPUT_CHANNELS_SUPPORTED 8
#define PCM_BLOCK_PER_CHANNEL_MS11 1536*2
#define AAC_BLOCK_PER_CHANNEL_MS11 768
#define NUMBER_BITS_IN_A_BYTE 8
#define AC3_BUFFER_SIZE 1920*2
#define MAX_OUTPUT_CHANNELS_SUPPORTED 8
#define PCM_2CH_OUT 0
#define PCM_MCH_OUT 1
#define SPDIF_OUT 2
#define COMPRESSED_OUT 2 // should be same as SPDIF_OUT
#define TRANSCODE_OUT 3
#define FACTOR_FOR_BUFFERING 2
#define NUM_DEVICES_SUPPORT_COMPR_DATA 2+1
#define NUM_SUPPORTED_CODECS 16
#define NUM_COLUMN_FOR_INDEXING 2
#define NUM_STATES_FOR_EACH_DEVICE_FMT 3
#define DECODER_TYPE_IDX 0
#define ROUTE_FORMAT_IDX 1
#define NUM_OF_PERIODS 8
enum {
LPCM,
MULTI_CH_PCM,
COMPR,
TRANSCODE
};
/*
List of indexes of the supported formats
Redundant formats such as (AAC-LC, HEAAC) are removed from the indexes as they
are treated with the AAC format
*/
enum {
PCM_IDX = 0,
AAC_IDX,
AC3_IDX,
EAC3_IDX,
DTS_IDX,
DTS_LBR_IDX,
MP3_IDX,
WMA_IDX,
WMA_PRO_IDX,
MP2_IDX,
ALL_FORMATS_IDX
};
/*
List of pass through's supported in the current usecases
*/
enum {
NO_PASSTHROUGH = 0,
AC3_PASSTHR,
EAC3_PASSTHR,
DTS_PASSTHR
};
/*
List of transcoder's supported in the current usecases
*/
enum {
NO_TRANSCODER = 0,
AC3_TRANSCODER,
DTS_TRANSCODER
};
/*
Requested end device format by user/app through set parameters
*/
enum {
UNCOMPRESSED = 0,
COMPRESSED,
COMPRESSED_CONVERT_EAC3_AC3,
COMPRESSED_CONVERT_ANY_AC3,
COMPRESSED_CONVERT_ANY_DTS,
AUTO_DEVICE_FORMAT,
UNCOMPRESSED_MCH, /* not to be exposed, internal use only */
COMPRESSED_CONVERT_AC3_ASSOC, /* not to be exposed, internal use only */
ALL_DEVICE_FORMATS
};
/*
List of type of data routed on end device
*/
typedef enum {
ROUTE_NONE = 0x0,
ROUTE_UNCOMPRESSED = 0x1,
ROUTE_COMPRESSED = 0x2,
ROUTE_SW_TRANSCODED = 0x10, //route sub-format, not to be used directly
ROUTE_DSP_TRANSCODED = 0x20, //route sub-format, not to be used directly
ROUTE_MCH = 0x40, //route sub-format, not to be used directly
ROUTE_UNCOMPRESSED_MCH = (ROUTE_UNCOMPRESSED | ROUTE_MCH),
ROUTE_SW_TRANSCODED_COMPRESSED = (ROUTE_COMPRESSED | ROUTE_SW_TRANSCODED),
ROUTE_DSP_TRANSCODED_COMPRESSED = (ROUTE_COMPRESSED | ROUTE_DSP_TRANSCODED)
}route_format_t;
/*
List of end device formats
*/
enum {
FORMAT_INVALID = -1,
FORMAT_PCM,
FORMAT_COMPR
};
/*
Below are the only different types of decode that we perform
*/
enum {
DSP_DECODE = 1, // render uncompressed
DSP_PASSTHROUGH = 2, // render compressed
DSP_TRANSCODE = 4, // render as compressed
SW_DECODE = 8, // render as uncompressed
SW_DECODE_MCH = 16, // render as uncompressed
SW_PASSTHROUGH = 32, // render compressed
SW_TRANSCODE = 64, // render compressed
NUM_DECODE_PATH = 7
};
/*
Modes of buffering that we can support
As of now, we only support input buffering to an extent specified by usecase
*/
enum {
NO_BUFFERING_MODE = 0,
INPUT_BUFFERING_MODE,
OUTPUT_BUFFEING_MODE
};
/*
playback controls
*/
enum {
PLAY = 1,
PAUSE = (1<<1),
RESUME = (1<<2),
SEEK = (1<<3),
EOS = (1<<4),
STOP = (1<<5),
STANDBY = (1<<6),
INIT = (1<<7),
};
/*
Multiple instance of use case
*/
enum {
STEREO_DRIVER = 0,
MULTI_CHANNEL_DRIVER,
COMRPESSED_DRIVER,
};
/*
Instance bits
*/
enum {
MULTI_CHANNEL_1_BIT = 1<<4,
MULTI_CHANNEL_2_BIT = 1<<5,
MULTI_CHANNEL_3_BIT = 1<<6,
COMPRESSED_1_BIT = 1<<12,
COMPRESSED_2_BIT = 1<<13,
COMPRESSED_3_BIT = 1<<14,
COMPRESSED_4_BIT = 1<<15,
COMPRESSED_5_BIT = 1<<16,
COMPRESSED_6_BIT = 1<<17
};
/*
List of support formats configured from frameworks
*/
static const int supportedFormats[NUM_SUPPORTED_CODECS] = {
AUDIO_FORMAT_PCM_16_BIT,
AUDIO_FORMAT_PCM_24_BIT,
AUDIO_FORMAT_AAC,
AUDIO_FORMAT_HE_AAC_V1,
AUDIO_FORMAT_HE_AAC_V2,
AUDIO_FORMAT_AAC_ADIF,
AUDIO_FORMAT_AC3,
AUDIO_FORMAT_AC3_DM,
AUDIO_FORMAT_EAC3,
AUDIO_FORMAT_EAC3_DM,
AUDIO_FORMAT_DTS,
AUDIO_FORMAT_DTS_LBR,
AUDIO_FORMAT_MP3,
AUDIO_FORMAT_WMA,
AUDIO_FORMAT_WMA_PRO,
AUDIO_FORMAT_MP2
};
/*
we can only have 6 types of decoder type stored with bit masks.
*/
static const int route_to_driver[NUM_DECODE_PATH][NUM_COLUMN_FOR_INDEXING] = {
{DSP_DECODE, ROUTE_UNCOMPRESSED_MCH},
{DSP_PASSTHROUGH, ROUTE_COMPRESSED},
{DSP_TRANSCODE, ROUTE_DSP_TRANSCODED_COMPRESSED},
{SW_DECODE, ROUTE_UNCOMPRESSED},
{SW_DECODE_MCH, ROUTE_UNCOMPRESSED_MCH},
{SW_PASSTHROUGH, ROUTE_COMPRESSED},
{SW_TRANSCODE, ROUTE_SW_TRANSCODED_COMPRESSED}
};
/*
table to query index based on the format
*/
static const int format_index[NUM_SUPPORTED_CODECS][NUM_COLUMN_FOR_INDEXING] = {
/*---------------------------------------------
| FORMAT | INDEX |
----------------------------------------------*/
{AUDIO_FORMAT_PCM_16_BIT, PCM_IDX},
{AUDIO_FORMAT_PCM_24_BIT, PCM_IDX},
{AUDIO_FORMAT_AAC, AAC_IDX},
{AUDIO_FORMAT_HE_AAC_V1, AAC_IDX},
{AUDIO_FORMAT_HE_AAC_V2, AAC_IDX},
{AUDIO_FORMAT_AAC_ADIF, AAC_IDX},
{AUDIO_FORMAT_AC3, AC3_IDX},
{AUDIO_FORMAT_AC3_DM, AC3_IDX},
{AUDIO_FORMAT_EAC3, EAC3_IDX},
{AUDIO_FORMAT_EAC3_DM, EAC3_IDX},
{AUDIO_FORMAT_DTS, DTS_IDX},
{AUDIO_FORMAT_DTS_LBR, DTS_LBR_IDX},
{AUDIO_FORMAT_MP3, MP3_IDX},
{AUDIO_FORMAT_WMA, WMA_IDX},
{AUDIO_FORMAT_WMA_PRO, WMA_PRO_IDX},
{AUDIO_FORMAT_MP2, MP2_IDX}
};
/*
Table to query non HDMI and SPDIF devices and their states such as type of
decode, type of data routed to end device and type of transcoding needed
*/
static const int usecase_decode_format[ALL_FORMATS_IDX*NUM_STATES_FOR_EACH_DEVICE_FMT] = {
/*-----------------
| UNCOMPR |
-----------------*/
/* PCM */
DSP_DECODE, //PCM_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
SW_DECODE, // AAC_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
SW_DECODE, //AC3_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
SW_DECODE, //EAC3_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
DSP_DECODE, //DTS_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
DSP_DECODE, //DTS_LBR_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
DSP_DECODE, //MP3_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
DSP_DECODE, //WMA_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
DSP_DECODE, //WMA_PRO_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER,//TRANSCODE_FORMAT
/* PCM */
DSP_DECODE, //MP2_IDX
FORMAT_PCM, //ROUTE_FORMAT
NO_TRANSCODER//TRANSCODE_FORMAT
};
/*
Table to query HDMI and SPDIF devices and their states such as type of
decode, type of data routed to end device and type of transcoding needed
*/
static const int usecase_docode_hdmi_spdif[ALL_FORMATS_IDX*NUM_STATES_FOR_EACH_DEVICE_FMT]
[ALL_DEVICE_FORMATS] = {
/*-------------------------------------------------------------------------------------------------------------------------------------------------------
| UNCOMPRESSED | COMPR | COMPR_CONV | COMPR_CONV | COMPR_CONV | AUTO | UNCOMPR_MCH | AC3_AC3 |
| | | EAC3_AC3 | ANY_AC3 | ANY_DTS | | | |
--------------------------------------------------------------------------------------------------------------------------------------------------------*/
/* PCM PCM PCM PCM PCM PCM PCM PCM */
{DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE|DSP_TRANSCODE, DSP_DECODE, DSP_DECODE, DSP_DECODE}, //PCM_IDX
{FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, DTS_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM PCM PCM AC3 PCM PCM PCM PCM */
{SW_DECODE, SW_DECODE, SW_DECODE, SW_TRANSCODE, DSP_DECODE, SW_DECODE, SW_DECODE_MCH, SW_DECODE}, //AAC_IDX
{FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, AC3_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM AC3 AC3 AC3 PCM AC3 PCM AC3 */
{SW_DECODE, SW_PASSTHROUGH, SW_PASSTHROUGH, SW_PASSTHROUGH, DSP_DECODE, SW_PASSTHROUGH, SW_DECODE_MCH, SW_TRANSCODE}, //AC3_IDX
{FORMAT_PCM, FORMAT_COMPR, FORMAT_COMPR, FORMAT_COMPR, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_COMPR}, //ROUTE_FORMAT
{NO_TRANSCODER, AC3_PASSTHR, AC3_PASSTHR, AC3_PASSTHR, NO_TRANSCODER, AC3_PASSTHR, NO_TRANSCODER, AC3_TRANSCODER}, //TRANSCODE_FMT
/* PCM EAC3 AC3 AC3 PCM EAC3 PCM PCM */
{SW_DECODE, SW_PASSTHROUGH, SW_TRANSCODE, SW_TRANSCODE, DSP_DECODE, SW_PASSTHROUGH, SW_DECODE_MCH, SW_TRANSCODE}, //EAC3_IDX
{FORMAT_PCM, FORMAT_COMPR, FORMAT_COMPR, FORMAT_COMPR, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_COMPR}, //ROUTE_FORMAT
{NO_TRANSCODER, EAC3_PASSTHR, AC3_TRANSCODER, AC3_TRANSCODER, NO_TRANSCODER, EAC3_PASSTHR, NO_TRANSCODER, AC3_TRANSCODER}, //TRANSCODE_FMT
/* PCM DTS PCM PCM DTS DTS PCM PCM */
{DSP_DECODE, DSP_PASSTHROUGH, DSP_DECODE, DSP_DECODE, DSP_PASSTHROUGH, DSP_PASSTHROUGH, DSP_DECODE, DSP_DECODE},//DTS_IDX
{FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, DTS_PASSTHR, NO_TRANSCODER, NO_TRANSCODER, DTS_PASSTHR, DTS_PASSTHR, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM DTS_LBR PCM PCM DTS DTS PCM PCM */
{DSP_DECODE, DSP_PASSTHROUGH, DSP_DECODE, DSP_DECODE, DSP_PASSTHROUGH, DSP_PASSTHROUGH, DSP_DECODE, DSP_DECODE},//DTS_LBR_IDX
{FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, DTS_PASSTHR, NO_TRANSCODER, NO_TRANSCODER, DTS_PASSTHR, DTS_PASSTHR, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM PCM PCM PCM DTS PCM PCM PCM */
{DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE|DSP_TRANSCODE, DSP_DECODE, DSP_DECODE, DSP_DECODE}, //MP3_IDX
{FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, DTS_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM PCM PCM PCM DTS PCM PCM PCM */
{DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE|DSP_TRANSCODE, DSP_DECODE, DSP_DECODE, DSP_DECODE}, //WMA_IDX
{FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, DTS_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM PCM PCM PCM DTS PCM PCM PCM */
{DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE|DSP_TRANSCODE, DSP_DECODE, DSP_DECODE, DSP_DECODE}, //WMA_PRO_IDX
{FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, DTS_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER}, //TRANSCODE_FMT
/* PCM PCM PCM PCM DTS PCM PCM PCM */
{DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE, DSP_DECODE|DSP_TRANSCODE, DSP_DECODE, DSP_DECODE, DSP_DECODE}, //MP2_IDX
{FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM, FORMAT_COMPR, FORMAT_PCM, FORMAT_PCM, FORMAT_PCM}, //ROUTE_FORMAT
{NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, DTS_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER, NO_TRANSCODER} //TRANSCODE_FMT
};
/*
List of decoders which require config as part of first buffer
*/
static const int decodersRequireConfig[] = {
AUDIO_FORMAT_AAC,
AUDIO_FORMAT_HE_AAC_V1,
AUDIO_FORMAT_HE_AAC_V2,
AUDIO_FORMAT_AAC_ADIF,
AUDIO_FORMAT_WMA,
AUDIO_FORMAT_WMA_PRO
};
/*
List of enum that are used in Broadcast.
NOTE: Need to be removed once broadcast is moved with updated states as above
*/
enum {
INVALID_FORMAT = -1,
PCM_FORMAT = 0,
COMPRESSED_FORMAT = 1,
COMPRESSED_FORCED_PCM_FORMAT = 2,
COMPRESSED_PASSTHROUGH_FORMAT = 3
};
struct audio_bitstream_sm {
int buffering_factor;
int buffering_factor_cnt;
// Buffer pointers for input and output
char *inp_buf;
char *inp_buf_curr_ptr;
char *inp_buf_write_ptr;
uint32_t inp_buf_size;
char *enc_out_buf;
char *enc_out_buf_write_ptr;
uint32_t enc_out_buf_size;
char *pcm_2_out_buf;
char *pcm_2_out_buf_write_ptr;
uint32_t pcm_2_out_buf_size;
char *pcm_mch_out_buf;
char *pcm_mch_out_buf_write_ptr;
uint32_t pcm_mch_out_buf_size;
char *passt_out_buf;
char *passt_out_buf_write_ptr;
uint32_t passt_out_buf_size;
};
/*
Meta data structure for handling compressed output
*/
struct output_metadata {
uint32_t metadataLength;
uint32_t bufferLength;
uint64_t timestamp;
uint32_t reserved[12];
};
#endif // QCOM_AUDIO_PLATFORM_H