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android / kernel / bcm / refs/tags/android-wear-6.0.1_r0.5 / . / sound / caph_hawaii / audio_driver / audio_vdriver.c
blob: a70ef6d8dfde0b2722c00bf8d9dcb09a1663a78d [file] [log] [blame]
/****************************************************************************
Copyright 2009 - 2012 Broadcom Corporation
Unless you and Broadcom execute a separate written software license agreement
governing use of this software, this software is licensed to you under the
terms of the GNU General Public License version 2 (the GPL), available at
http://www.broadcom.com/licenses/GPLv2.php
with the following added to such license:
As a special exception, the copyright holders of this software give you
permission to link this software with independent modules, and to copy and
distribute the resulting executable under terms of your choice, provided
that you also meet, for each linked independent module, the terms and
conditions of the license of that module.
An independent module is a module which is not derived from this software.
The special exception does not apply to any modifications of the software.
Notwithstanding the above, under no circumstances may you combine this software
in any way with any other Broadcom software provided under a license other than
the GPL, without Broadcom's express prior written consent.
***************************************************************************/
/**
*
* @file audio_vdriver.c
* @brief Audio VDriver API for CAPH
*
******************************************************************************/
/*=============================================================================
// Include directives
//=============================================================================
*/
#include <linux/kernel.h>
#include <linux/slab.h>
#include <linux/completion.h>
#include "mobcom_types.h"
#include "audio_consts.h"
#include "csl_apcmd.h"
#include "bcm_fuse_sysparm_CIB.h"
#include "csl_caph.h"
#include "csl_dsp.h"
#include "csl_caph_audioh.h"
#include "resultcode.h"
#include "csl_caph_hwctrl.h"
#include "audio_vdriver.h"
#include "audio_rpc.h"
#include <mach/comms/platform_mconfig.h>
#include "io.h"
#include "csl_dsp_cneon_api.h"
#if defined(ENABLE_DMA_VOICE)
#include "csl_dsp_caph_control_api.h"
#endif
#include "audio_controller.h"
#include "audio_ddriver.h"
#include "audio_trace.h"
#include "taskmsgs.h"
#include "ipcproperties.h"
#include "rpc_global.h"
#include "rpc_ipc.h"
#include "xdr_porting_layer.h"
#include "xdr.h"
#include "rpc_api.h"
/**
*
* @addtogroup AudioDriverGroup
* @{
*/
/*=============================================================================
// Private Type and Constant declarations
//=============================================================================
*/
typedef void (*AUDDRV_User_CB) (UInt32 param1, UInt32 param2, UInt32 param3);
struct _AUDDRV_PathID_t {
CSL_CAPH_PathID ulPathID;
CSL_CAPH_PathID ul2PathID;
CSL_CAPH_PathID dlPathID;
};
#define AUDDRV_PathID_t struct _AUDDRV_PathID_t
/*=============================================================================
// Private Variable declarations
//=============================================================================
*/
#if defined(CONFIG_BCM_MODEM)
#define AUDIO_MODEM(a) a
#else
#define AUDIO_MODEM(a)
#endif
/*this is needed because DSPCMD_AUDIO_ENABLE sets/clears AMCR.
AUDEN for both voiceIn and voiceOut */
static Boolean voicePlayOutpathEnabled = FALSE;
static AUDDRV_PathID_t telephonyPathID;
static void *sUserCB;
static CSL_CAPH_DEVICE_e sink = CSL_CAPH_DEV_NONE;
static Boolean userEQOn = FALSE;
/* If TRUE, bypass EQ filter setting request from audio controller.*/
/* used in pcm i/f control. assume one mic, one spkr. */
static AUDIO_SOURCE_Enum_t currVoiceMic = AUDIO_SOURCE_UNDEFINED;
static AUDIO_SINK_Enum_t currVoiceSpkr = AUDIO_SINK_UNDEFINED;
static Boolean dspECEnable = TRUE;
static Boolean dspNSEnable = TRUE;
static Boolean controlFlagForCustomGain = FALSE;
static Boolean inCallRateChange = FALSE;
static int audio_tuning_flag;
static Boolean voiceRecOn = FALSE;
static AUDIO_SOURCE_Enum_t voiceInMic;
static AUDIO_SAMPLING_RATE_t voiceInSr;
struct _Audio_Driver_t {
UInt8 isRunning;
UInt32 taskID;
};
#define Audio_Driver_t struct _Audio_Driver_t
static Audio_Driver_t sAudDrv = { 0 };
static audio_codecId_handler_t codecId_handler;
static audio_handleCPReset_handler_t cpReset_handler;
static Int32 curCallMode = CALL_MODE_NONE;
static UInt32 curCallSampleRate = AUDIO_SAMPLING_RATE_8000;
static int isIHFDL48k = 1; /*set to enable 48k IHF call DL*/
static int echo_ref_mic;
struct completion audioEnableDone;
struct completion arm2spHqDLInitDone;
struct AUDCTRL_SPKR_Mapping_t {
AUDIO_SINK_Enum_t spkr;
CSL_CAPH_DEVICE_e dev;
};
/* must match AUDIO_SINK_Enum_t */
static struct AUDCTRL_SPKR_Mapping_t \
SPKR_Mapping_Table[AUDIO_SINK_TOTAL_COUNT] = {
/* sink ino Device ID */
[AUDIO_SINK_HANDSET] = {AUDIO_SINK_HANDSET, CSL_CAPH_DEV_EP},
[AUDIO_SINK_HEADSET] = {AUDIO_SINK_HEADSET, CSL_CAPH_DEV_HS},
[AUDIO_SINK_HANDSFREE] = {AUDIO_SINK_HANDSFREE, CSL_CAPH_DEV_IHF},
[AUDIO_SINK_BTM] = {AUDIO_SINK_BTM, CSL_CAPH_DEV_BT_SPKR},
[AUDIO_SINK_LOUDSPK] = {AUDIO_SINK_LOUDSPK, CSL_CAPH_DEV_IHF},
[AUDIO_SINK_TTY] = {AUDIO_SINK_TTY, CSL_CAPH_DEV_HS},
[AUDIO_SINK_HAC] = {AUDIO_SINK_HAC, CSL_CAPH_DEV_EP},
[AUDIO_SINK_USB] = {AUDIO_SINK_USB, CSL_CAPH_DEV_MEMORY},
[AUDIO_SINK_BTS] = {AUDIO_SINK_BTS, CSL_CAPH_DEV_BT_SPKR},
[AUDIO_SINK_I2S] = {AUDIO_SINK_I2S, CSL_CAPH_DEV_FM_TX},
[AUDIO_SINK_VIBRA] = {AUDIO_SINK_VIBRA, CSL_CAPH_DEV_VIBRA},
[AUDIO_SINK_HEADPHONE] = {AUDIO_SINK_HEADPHONE, CSL_CAPH_DEV_HS},
[AUDIO_SINK_VALID_TOTAL] = {AUDIO_SINK_VALID_TOTAL, CSL_CAPH_DEV_NONE},
[AUDIO_SINK_MEM] = {AUDIO_SINK_MEM, CSL_CAPH_DEV_MEMORY},
[AUDIO_SINK_DSP] = {AUDIO_SINK_DSP, CSL_CAPH_DEV_DSP},
[AUDIO_SINK_UNDEFINED] = {AUDIO_SINK_UNDEFINED, CSL_CAPH_DEV_NONE}
};
struct AUDIO_SOURCE_Mapping_t {
AUDIO_SOURCE_Enum_t mic;
CSL_CAPH_DEVICE_e dev;
};
/* must match AUDIO_SOURCE_Enum_t */
static struct AUDIO_SOURCE_Mapping_t \
MIC_Mapping_Table[AUDIO_SOURCE_TOTAL_COUNT] = {
/* source info Device ID */
{AUDIO_SOURCE_UNDEFINED, CSL_CAPH_DEV_NONE},
{AUDIO_SOURCE_ANALOG_MAIN, CSL_CAPH_DEV_ANALOG_MIC},
{AUDIO_SOURCE_ANALOG_AUX, CSL_CAPH_DEV_HS_MIC},
{AUDIO_SOURCE_DIGI1, CSL_CAPH_DEV_DIGI_MIC_R},
{AUDIO_SOURCE_DIGI2, CSL_CAPH_DEV_DIGI_MIC_L},
{AUDIO_SOURCE_DIGI3, CSL_CAPH_DEV_EANC_DIGI_MIC_R},
{AUDIO_SOURCE_DIGI4, CSL_CAPH_DEV_EANC_DIGI_MIC_L},
{AUDIO_SOURCE_MIC_ARRAY1, CSL_CAPH_DEV_DIGI_MIC},
{AUDIO_SOURCE_MIC_ARRAY2, CSL_CAPH_DEV_EANC_DIGI_MIC},
{AUDIO_SOURCE_BTM, CSL_CAPH_DEV_BT_MIC},
{AUDIO_SOURCE_USB, CSL_CAPH_DEV_MEMORY},
{AUDIO_SOURCE_I2S, CSL_CAPH_DEV_FM_RADIO},
{AUDIO_SOURCE_EP, CSL_CAPH_DEV_EP},
{AUDIO_SOURCE_IHF, CSL_CAPH_DEV_IHF},
{AUDIO_SOURCE_HS, CSL_CAPH_DEV_HS},
{AUDIO_SOURCE_RESERVED1, CSL_CAPH_DEV_NONE},
{AUDIO_SOURCE_RESERVED2, CSL_CAPH_DEV_NONE},
{AUDIO_SOURCE_VALID_TOTAL, CSL_CAPH_DEV_NONE},
{AUDIO_SOURCE_SPEECH_DIGI, CSL_CAPH_DEV_DIGI_MIC},
{AUDIO_SOURCE_MEM, CSL_CAPH_DEV_MEMORY},
{AUDIO_SOURCE_DSP, CSL_CAPH_DEV_DSP}
};
struct AUDCTRL_SPKR_MIC_Mapping_t {
AUDIO_SINK_Enum_t spkr;
AUDIO_SOURCE_Enum_t pri_mic;
AUDIO_SOURCE_Enum_t sec_mic;
};
/* must match AUDIO_SINK_Enum_t */
static struct AUDCTRL_SPKR_MIC_Mapping_t \
SPKR_MIC_Mapping_Table[AUDIO_SINK_TOTAL_COUNT] = {
/* sink info primary MIC ID Secondary MIC ID */
[AUDIO_SINK_HANDSET] = {AUDIO_SINK_HANDSET,
AUDIO_SOURCE_ANALOG_MAIN, AUDIO_SOURCE_DIGI2},
[AUDIO_SINK_HEADSET] = {AUDIO_SINK_HEADSET,
AUDIO_SOURCE_ANALOG_AUX, AUDIO_SOURCE_DIGI2},
[AUDIO_SINK_HANDSFREE] = {AUDIO_SINK_HANDSFREE,
AUDIO_SOURCE_ANALOG_MAIN, AUDIO_SOURCE_DIGI2},
[AUDIO_SINK_BTM] = {AUDIO_SINK_BTM,
AUDIO_SOURCE_BTM, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_LOUDSPK] = {AUDIO_SINK_LOUDSPK,
AUDIO_SOURCE_ANALOG_MAIN, AUDIO_SOURCE_DIGI2},
[AUDIO_SINK_TTY] = {AUDIO_SINK_TTY,
AUDIO_SOURCE_ANALOG_AUX, AUDIO_SOURCE_DIGI2},
[AUDIO_SINK_HAC] = {AUDIO_SINK_HAC,
AUDIO_SOURCE_ANALOG_MAIN, AUDIO_SOURCE_DIGI2},
[AUDIO_SINK_USB] = {AUDIO_SINK_USB,
AUDIO_SOURCE_USB, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_BTS] = {AUDIO_SINK_BTS,
AUDIO_SOURCE_BTM, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_I2S] = {AUDIO_SINK_I2S,
AUDIO_SOURCE_UNDEFINED, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_VIBRA] = {AUDIO_SINK_VIBRA,
AUDIO_SOURCE_UNDEFINED, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_HEADPHONE] = {AUDIO_SINK_HEADPHONE,
AUDIO_SOURCE_ANALOG_MAIN, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_VALID_TOTAL] = {AUDIO_SINK_VALID_TOTAL,
AUDIO_SOURCE_UNDEFINED, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_MEM] = {AUDIO_SINK_MEM,
AUDIO_SOURCE_UNDEFINED, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_DSP] = {AUDIO_SINK_DSP,
AUDIO_SOURCE_UNDEFINED, AUDIO_SOURCE_UNDEFINED},
[AUDIO_SINK_UNDEFINED] = {AUDIO_SINK_UNDEFINED,
AUDIO_SOURCE_UNDEFINED, AUDIO_SOURCE_UNDEFINED}
};
/*=============================================================================
// Private function prototypes
//=============================================================================
*/
#if (!defined(CONFIG_BCM_MODEM) || defined(JAVA_ZEBU_TEST))
/* this array initializes the system parameters with the values
from param_audio_rhea.txt for different modes(mic_pga - hw_sidetone_gain)*/
static AudioSysParm_t audio_parm_table[AUDIO_MODE_NUMBER_VOICE] = {
{160, 36, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 1,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 65520,
65520, 1, 0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{160, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 16, 16,
0, 0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 65520,
65520, 0, 0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 16, 16, 0,
1, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 65520,
65520, 1, 0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 8, 8, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 65520,
65520, 0, 0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} },
{72, 36, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0, 96, 0, 0, 0, 0,
0, 0, {0}, {0}, {0} }
};
static SysMultimediaAudioParm_t mmaudio_parm_table
[AUDIO_APP_MM_NUMBER*AUDIO_MODE_NUMBER] = {
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} },
{7, 72, 36, 24, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 96, 0, 0,
96, 0, 0, 0, 0, {-4200, -3900, -3600, -3300, -3000, -2700,
-2400, -2100, -1800, -1500, 1200, -900, -600, -300, 0} }
};
#endif
#if defined(CONFIG_BCM_MODEM) && (!defined(JAVA_ZEBU_TEST))
SysAudioParm_t *AudParmP(void)
#else
AudioSysParm_t *AudParmP(void)
#endif
{
#if defined(CONFIG_BCM_MODEM) && (!defined(JAVA_ZEBU_TEST))
return APSYSPARM_GetAudioParmAccessPtr();
#else
return &audio_parm_table[0];
#endif
}
SysMultimediaAudioParm_t *MMAudParmP(void)
{
#if defined(CONFIG_BCM_MODEM) && !defined(JAVA_ZEBU_TEST)
return APSYSPARM_GetMultimediaAudioParmAccessPtr();
#else
return &mmaudio_parm_table[0];
#endif
}
static UInt32 *AUDIO_GetIHF48KHzBufferBaseAddress(void);
static void AUDDRV_Telephony_InitHW(AUDIO_SOURCE_Enum_t mic,
AUDIO_SINK_Enum_t speaker,
AUDIO_SAMPLING_RATE_t sample_rate,
Boolean bNeedDualMic);
static void AUDDRV_HW_SetFilter(AUDDRV_HWCTRL_FILTER_e filter,
void *coeff);
static void AUDDRV_HW_EnableSideTone(AudioMode_t audio_mode);
static void AUDDRV_HW_DisableSideTone(AudioMode_t audio_mode);
#ifdef CONFIG_BCM_MODEM
static void AP_ProcessAudioEnableDone(UInt16 enabled_path);
static void AP_ProcessArm2spHqDLInitDone(void);
#endif
/*=============================================================================
// Functions
//=============================================================================
*/
/*=============================================================================
//
// Function Name: AUDDRV_Init
//
// Description: Inititialize audio system
//
//=============================================================================
*/
void AUDDRV_Init(void)
{
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_Init\n");
if (sAudDrv.isRunning == TRUE)
return;
#ifdef CONFIG_BCM_MODEM
/* register DSP VPU status processing handlers */
AUDIO_MODEM(CSL_RegisterVPUCaptureStatusHandler
((VPUCaptureStatusCB_t) &VPU_Capture_Request);)
#endif
#if 0 /*These features are not needed in LMP now. */
AUDIO_MODEM(CSL_RegisterVPURenderStatusHandler
((VPURenderStatusCB_t) &VPU_Render_Request);)
AUDIO_MODEM(CSL_RegisterUSBStatusHandler
((USBStatusCB_t) &AUDDRV_USB_HandleDSPInt);)
#endif
AUDIO_MODEM(CSL_RegisterVoIPStatusHandler
((VoIPStatusCB_t) &VOIP_ProcessVOIPDLDone);)
AUDIO_MODEM(CSL_RegisterMainAMRStatusHandler
((MainAMRStatusCB_t) &AP_ProcessStatusMainAMRDone);)
AUDIO_MODEM(CSL_RegisterARM2SPRenderStatusHandler
((ARM2SPRenderStatusCB_t) &ARM2SP_Render_Request);)
AUDIO_MODEM(CSL_RegisterARM2SP2RenderStatusHandler
((ARM2SP2RenderStatusCB_t) &ARM2SP2_Render_Request);)
AUDIO_MODEM(CSL_RegisterAudioLogHandler
((AudioLogStatusCB_t) &AUDLOG_ProcessLogChannel);)
AUDIO_MODEM(CSL_RegisterVOIFStatusHandler
((VOIFStatusCB_t) &VOIF_Buffer_Request);)
AUDIO_MODEM(CSL_RegisterAudioEnableDoneHandler
((AudioEnableDoneStatusCB_t)
&AP_ProcessAudioEnableDone);)
AUDIO_MODEM(CSL_RegisterARM2SP_HQ_DL_InitDoneHandler
((ARM2SP_HQ_DL_InitDoneStatusCB_t)
&AP_ProcessArm2spHqDLInitDone);)
Audio_InitRpc();
sAudDrv.isRunning = TRUE;
init_completion(&audioEnableDone);
init_completion(&arm2spHqDLInitDone);
}
/**********************************************************************
//
// Shutdown audio driver task
//
// @return void
// @note
**********************************************************************/
void AUDDRV_Shutdown(void)
{
if (sAudDrv.isRunning == FALSE)
return;
sAudDrv.isRunning = FALSE;
}
/*=============================================================================
//
// Function Name: AUDDRV_Telephony_Init
//
// Description: Initialize audio system for voice call
//
//=============================================================================
*/
/*Prepare DSP before turn on hardware audio path for voice call.
// This is part of the control sequence for starting telephony audio.*/
void AUDDRV_Telephony_Init(AUDIO_SOURCE_Enum_t mic, AUDIO_SINK_Enum_t speaker,
AudioMode_t mode, AudioApp_t app, int bNeedDualMic,
int bmuteVoiceCall)
{
#if defined(ENABLE_DMA_VOICE)
UInt16 dma_mic_spk;
#endif
Boolean ec_enable_from_sysparm = dspECEnable;
Boolean ns_enable_from_sysparm = dspNSEnable;
/*-/////////////////////////////////////////////////////////////////
// Phone Setup Sequence
// 1. Init CAPH HW
// 2. Send DSP command DSPCMD_TYPE_AUDIO_ENABLE
// 3. If requires 48KHz for example in IHF mode,
Send VPRIPCMDQ_ENABLE_48KHZ_SPEAKER_OUTPUT
////////////////////////////////////////////////////////////////-*/
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_Telephony_Init\n");
csl_caph_ControlHWClock(TRUE); /*enable clock before any DSP command*/
currVoiceMic = mic;
currVoiceSpkr = speaker;
#if defined(CONFIG_MACH_HAWAII_GARNET) || defined(CONFIG_MACH_JAVA_GARNET)
/*For garnet need to enable regulator to power on analog switch which
selects IHF protection loopback or Analog Mic line to BB*/
if (currVoiceMic == AUDIO_SOURCE_ANALOG_MAIN)
csl_ControlHW_dmic_regulator(TRUE);
#endif
#ifdef CONFIG_AUDIO_S2
/* use the main mic for handset mode,
and use sub mic for the loud speaker*/
if (speaker == AUDIO_SINK_HANDSET)
(*((UInt32 *)(HW_IO_PHYS_TO_VIRT(0x35004844)))
= (UInt32)0x00000343);
else if (speaker == AUDIO_SINK_LOUDSPK)
(*((UInt32 *)(HW_IO_PHYS_TO_VIRT(0x35004844)))
= (UInt32)0x00000303);
#endif
audio_control_dsp(AUDDRV_DSPCMD_MUTE_DSP_UL, 0, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_EC_NS_ON, FALSE, FALSE, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_DUAL_MIC_ON, FALSE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_TURN_UL_COMPANDEROnOff, FALSE, 0,
0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_UL, FALSE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, FALSE, 0, 0, 0, 0);
AUDDRV_Telephony_InitHW(mic, speaker, curCallSampleRate, bNeedDualMic);
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_set_samp_rate(curCallSampleRate);
#endif
/* Set new filter coef, sidetone filters, gains. */
AUDDRV_SetAudioMode(mode, app,
telephonyPathID.ulPathID,
telephonyPathID.ul2PathID,
telephonyPathID.dlPathID);
if (speaker == AUDIO_SINK_LOUDSPK && isIHFDL48k == 1) {
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_disable_path((UInt16)
(DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN));
/* dma_mic_spk = (UInt16) DSP_AADMAC_PRI_MIC_EN;*/
dma_mic_spk = ((UInt16)(DSP_AADMAC_PRI_MIC_EN)) | ((UInt16)
(DSP_AADMAC_48KHZ_SPKR_EN)) |
((UInt16)DSP_AADMAC_RETIRE_DS_CMD);
if (bNeedDualMic)
dma_mic_spk |= (UInt16) (DSP_AADMAC_SEC_MIC_EN);
csl_dsp_caph_control_aadmac_enable_path(dma_mic_spk);
#endif
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, TRUE, 0, 0, 0, 0);
} else {
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_disable_path((UInt16)
(DSP_AADMAC_48KHZ_SPKR_EN) |
(UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN));
dma_mic_spk =
(UInt16) (DSP_AADMAC_PRI_MIC_EN) |
(UInt16) (DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_RETIRE_DS_CMD);
if (bNeedDualMic)
dma_mic_spk |= (UInt16) (DSP_AADMAC_SEC_MIC_EN);
if (speaker == AUDIO_SINK_BTM)
dma_mic_spk |= (UInt16)
(DSP_AADMAC_PACKED_16BIT_IN_OUT_EN);
csl_dsp_caph_control_aadmac_enable_path(dma_mic_spk);
#endif
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, TRUE, 0, 0, 0, 0);
}
if (curCallMode != PTT_CALL) {
#if defined(ENABLE_DMA_VOICE)
audio_control_dsp(
AUDDRV_DSPCMD_AUDIO_CONNECT_DL, TRUE, 0, 0, 0, 0);
#else
audio_control_dsp(
AUDDRV_DSPCMD_AUDIO_CONNECT_DL, TRUE,
AUDCTRL_Telephony_HW_16K(mode), 0, 0, 0);
#endif
} else
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_Telephony_Init: "
"PTT, skip AUDIO_CONNECT_DL\n");
#if defined(ENABLE_DMA_VOICE)
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_UL, TRUE, 0, 0, 0, 0);
#else
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_UL, TRUE,
AUDCTRL_Telephony_HW_16K(mode), 0, 0, 0);
#endif
/* audio_control_dsp(DSPCMD_TYPE_EC_NS_ON, TRUE, TRUE, 0, 0, 0); */
audio_control_dsp(AUDDRV_DSPCMD_EC_NS_ON, ec_enable_from_sysparm,
ns_enable_from_sysparm, 0, 0, 0);
if (bNeedDualMic == TRUE)
audio_control_dsp(AUDDRV_DSPCMD_DUAL_MIC_ON, TRUE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_TURN_UL_COMPANDEROnOff, TRUE, 0,
0, 0, 0);
if (bmuteVoiceCall == FALSE)
audio_control_dsp(AUDDRV_DSPCMD_UNMUTE_DSP_UL, 0, 0, 0, 0, 0);
/* per call basis: enable the DTX by calling stack api when call connected */
audio_control_generic(AUDDRV_CPCMD_ENABLE_DSP_DTX, TRUE, 0, 0, 0, 0);
return;
}
/*=============================================================================
//
// Function Name: AUDDRV_Telephony_RateChange
//
// Description: Change the sample rate for voice call
//
//=============================================================================
*/
void AUDDRV_Telephony_RateChange(AudioMode_t mode,
AudioApp_t audio_app, int bNeedDualMic, int bmuteVoiceCall)
{
inCallRateChange = TRUE;
AUDDRV_Telephony_Deinit();
AUDDRV_Telephony_Init(currVoiceMic, currVoiceSpkr, mode, audio_app,
bNeedDualMic, bmuteVoiceCall);
inCallRateChange = FALSE;
return;
}
/*********************************************************************
//
// Registers callback for rate change request
//
// @param callback function
// @return void
// @note
**********************************************************************/
void AUDDRV_RegisterRateChangeCallback(audio_codecId_handler_t codecId_cb)
{
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_RegisterRateChangeCallback, 0x%lx\n",
(long unsigned int)codecId_cb);
codecId_handler = codecId_cb;
}
/*********************************************************************
//
// Registers callback for handling cp reset
//
// @param callback function
// @return void
// @note
**********************************************************************/
void AUDDRV_RegisterHandleCPResetCB(
audio_handleCPReset_handler_t cpReset_cb)
{
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_RegisterHandleCPResetCB");
cpReset_handler = cpReset_cb;
}
/*********************************************************************
//
// Function Name: AUDDRV_EC
//
// Description: DSP Echo cancellation ON/OFF
//
**********************************************************************/
void AUDDRV_EC(Boolean enable, UInt32 arg)
{
dspECEnable = enable;
}
/*********************************************************************
//
// Function Name: AUDDRV_NS
//
// Description: DSP Noise Suppression ON/OFF
//
**********************************************************************/
void AUDDRV_NS(Boolean enable)
{
dspNSEnable = enable;
}
/****************************************************************************
*
* Function Name: AUDDRV_ECreset_NLPoff
*
* Description: DSP control to turn EC ON but NLP OFF, regardless of sysparm
*
****************************************************************************/
void AUDDRV_ECreset_NLPoff(Boolean ECenable)
{
audio_control_dsp(AUDDRV_DSPCMD_CONTROL_ECHO_CANCELLATION,
ECenable, 0, 0, 0, 0);
}
/*********************************************************************
//
// Post rate change message for telephony session
//
// @param codec ID
// @return void
// @note
**********************************************************************/
void AUDDRV_Telephone_RequestRateChange(int codecID)
{
if (codecId_handler != NULL)
codecId_handler(codecID);
}
/*********************************************************************
//
// Post cp reset message
//
// @param none
// @return void
// @note
**********************************************************************/
void AUDDRV_HandleCPReset(Boolean cp_reset_start)
{
if (cpReset_handler != NULL)
cpReset_handler(cp_reset_start);
}
/*=============================================================================
//
// Function Name: AUDDRV_Telephony_Deinit
//
// Description: DeInitialize audio system for voice call
//
//=============================================================================
*/
/*Prepare DSP before turn off hardware audio path for voice call.
// This is part of the control sequence for ending telephony audio.*/
void AUDDRV_Telephony_Deinit(void)
{
#if defined(ENABLE_DMA_VOICE)
UInt16 dma_mic_spk;
#endif
aTrace(LOG_AUDIO_DRIVER, "%s voiceOut %d voiceRecOn %d\n",
__func__, voicePlayOutpathEnabled, voiceRecOn);
/* continues speech playback when end the phone call.
// continues speech recording when end the phone call.*/
/* if ( FALSE==vopath_enabled && FALSE==DspVoiceIfActive_DL()
&& FALSE==vipath_enabled )
{ */
/* a quick fix not to disable voice path for speech playbck
when end the phone call. */
/* disable the DTX by calling stack api when call disconnected */
if (voicePlayOutpathEnabled == FALSE) {
audio_control_generic(AUDDRV_CPCMD_ENABLE_DSP_DTX, FALSE, 0, 0,
0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, FALSE, 0, 0,
0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_UL, FALSE, 0, 0,
0, 0);
audio_control_dsp(AUDDRV_DSPCMD_EC_NS_ON, FALSE, FALSE, 0, 0,
0);
audio_control_dsp(AUDDRV_DSPCMD_DUAL_MIC_ON, FALSE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_TURN_UL_COMPANDEROnOff,
FALSE, 0, 0, 0, 0);
#if defined(ENABLE_DMA_VOICE)
dma_mic_spk =
(UInt16) (DSP_AADMAC_PRI_MIC_EN) |
(UInt16) (DSP_AADMAC_48KHZ_SPKR_EN) |
(UInt16) (DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_SEC_MIC_EN) |
(UInt16) (DSP_AADMAC_ECHO_REF_EN) |
(UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN);
csl_dsp_caph_control_aadmac_disable_path(dma_mic_spk);
#endif
audio_control_dsp(AUDDRV_DSPCMD_MUTE_DSP_UL, 0, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, FALSE,
0, 0, 0, 0);
AUDDRV_Telephony_DeinitHW();
}
#if defined(CONFIG_MACH_HAWAII_GARNET) || defined(CONFIG_MACH_JAVA_GARNET)
/*For garnet need to disable regulator to power off analog switch
which selects IHF protection loopback or Analog Mic line to BB*/
if (currVoiceMic == AUDIO_SOURCE_ANALOG_MAIN)
csl_ControlHW_dmic_regulator(FALSE);
#endif
if (!inCallRateChange) {
currVoiceMic = AUDIO_SOURCE_UNDEFINED;
currVoiceSpkr = AUDIO_SINK_UNDEFINED;
}
if (voiceRecOn)
AUDDRV_EnableDSPInput(voiceInMic, voiceInSr);
return;
}
/*=============================================================================
//
// Function Name: AUDDRV_EnableDSPOutput
//
// Description: Enable audio DSP output for voice call
//
//=============================================================================
*/
void AUDDRV_EnableDSPOutput(AUDIO_SINK_Enum_t sink,
AUDIO_SAMPLING_RATE_t sample_rate)
{
aTrace(LOG_AUDIO_DRIVER, "%s mixer %d, sample_rate %u\n",
__func__, sink, sample_rate);
usleep_range(5000, 10000);
/* sometimes BBC video has no audio.
This delay may help the mixer filter and mixer gain loading. */
currVoiceSpkr = sink;
if (sample_rate == AUDIO_SAMPLING_RATE_8000) {
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_set_samp_rate
(AUDIO_SAMPLING_RATE_8000);
if (sink == AUDIO_SINK_BTM)
csl_dsp_caph_control_aadmac_enable_path((UInt16)
(DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN) |
(UInt16) (DSP_AADMAC_RETIRE_DS_CMD));
else
csl_dsp_caph_control_aadmac_enable_path((UInt16)
(DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_RETIRE_DS_CMD));
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE,
1, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, 1, 0, 0,
0, 0);
#else
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, 1, 0, 0, 0,
0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, 1, 0, 0,
0, 0);
#endif
} else {
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_set_samp_rate
(AUDIO_SAMPLING_RATE_16000);
if (sink == AUDIO_SINK_BTM)
csl_dsp_caph_control_aadmac_enable_path((UInt16)
(DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN) |
(UInt16) (DSP_AADMAC_RETIRE_DS_CMD));
else
csl_dsp_caph_control_aadmac_enable_path((UInt16)
(DSP_AADMAC_SPKR_EN) |
(UInt16) (DSP_AADMAC_RETIRE_DS_CMD));
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE,
1, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, 1, 0, 0,
0, 0);
#else
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, 1, 1, 0, 0,
0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, 1, 1, 0,
0, 0);
#endif
}
voicePlayOutpathEnabled = TRUE;
aTrace(LOG_AUDIO_DRIVER, "%s voicePlayOutpathEnabled=%d",
__func__, voicePlayOutpathEnabled);
}
/* ============================================================================
//
// Function Name: AUDDRV_DisableDSPOutput
//
// Description: Disable audio DSP output for playback
//
//=============================================================================
*/
void AUDDRV_DisableDSPOutput(void)
{
aTrace(LOG_AUDIO_DRIVER, "%s", __func__);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, FALSE, 0, 0, 0,
0);
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_disable_path(((UInt16)
DSP_AADMAC_SPKR_EN) | ((UInt16)(DSP_AADMAC_48KHZ_SPKR_EN)) |
((UInt16)(DSP_AADMAC_PACKED_16BIT_IN_OUT_EN)));
#endif
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, FALSE, 0, 0, 0, 0);
voicePlayOutpathEnabled = FALSE;
}
/*=============================================================================
//
// Function Name: AUDDRV_EnableDSPInput
//
// Description: Enable audio DSP input for voice record
//
//=============================================================================
*/
void AUDDRV_EnableDSPInput(AUDIO_SOURCE_Enum_t source,
AUDIO_SAMPLING_RATE_t sample_rate)
{
#if defined(ENABLE_DMA_VOICE)
UInt16 dma_mic_spk;
#endif
UInt32 flag16k = 0;
aTrace(LOG_AUDIO_DRIVER, "%s source %d voiceRecOn %d, ulPath %d\n",
__func__, source, voiceRecOn, telephonyPathID.ulPathID);
voiceRecOn = TRUE;
voiceInMic = source;
voiceInSr = sample_rate;
/*only in non-call mode, tell DSP to start UL*/
if (telephonyPathID.ulPathID)
return;
#ifdef CONFIG_BCM_MODEM
csl_dsp_caph_control_aadmac_set_samp_rate(sample_rate);
#endif
if (sample_rate == AUDIO_SAMPLING_RATE_16000)
flag16k = 1;
#if defined(ENABLE_DMA_VOICE)
dma_mic_spk =
(UInt16) (DSP_AADMAC_PRI_MIC_EN) |
(UInt16) (DSP_AADMAC_RETIRE_DS_CMD);
if (echo_ref_mic)
dma_mic_spk |= (UInt16) (DSP_AADMAC_ECHO_REF_EN);
if (source == AUDIO_SOURCE_BTM)
dma_mic_spk |= (UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN);
csl_dsp_caph_control_aadmac_enable_path(dma_mic_spk);
#endif
/* DSPCMD_TYPE_AUDIO_CONNECT should be called after DSPCMD_TYPE_AUDIO_ENABLE */
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, 1, flag16k, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_UL, 1, flag16k, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_EC_NS_ON, TRUE, TRUE, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_DUAL_MIC_ON, TRUE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_TURN_UL_COMPANDEROnOff, TRUE, 0,
0, 0, 0);
/*
When voice call ends, DSPCMD_TYPE_MUTE_DSP_UL is being sent to DSP and
this command mutes UL record gain
Not sure why this is done and there is no clarification if we really
need to send the MUTE UL command when disconnecting the voice call.
For now, when voice record is started, UMUTE UL command will be sent */
audio_control_dsp(AUDDRV_DSPCMD_UNMUTE_DSP_UL, 0, 0, 0, 0, 0);
currVoiceMic = source;
}
/*=============================================================================
//
// Function Name: AUDDRV_DisableDSPInput
//
// Description: Disable audio DSP input for record
//
//=============================================================================
*/
void AUDDRV_DisableDSPInput(int stop)
{
#if defined(ENABLE_DMA_VOICE)
UInt16 dma_mic_spk;
#endif
aTrace(LOG_AUDIO_DRIVER, "%s stop %d, voiceRecOn %d, ulPath %d\n",
__func__, stop, voiceRecOn, telephonyPathID.ulPathID);
if (stop)
voiceRecOn = FALSE;
/*only in non-call mode, tell DSP to stop UL*/
if (telephonyPathID.ulPathID)
return;
audio_control_dsp(AUDDRV_DSPCMD_EC_NS_ON, FALSE, FALSE, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_UL, FALSE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_DUAL_MIC_ON, FALSE, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_TURN_UL_COMPANDEROnOff,
FALSE, 0, 0, 0, 0);
#ifdef CONFIG_BCM_MODEM
dma_mic_spk =
(UInt16) (DSP_AADMAC_PRI_MIC_EN) |
(UInt16) (DSP_AADMAC_SEC_MIC_EN) |
(UInt16) (DSP_AADMAC_ECHO_REF_EN) |
(UInt16) (DSP_AADMAC_PACKED_16BIT_IN_OUT_EN);
#endif
#if defined(ENABLE_DMA_VOICE)
csl_dsp_caph_control_aadmac_disable_path(dma_mic_spk);
#endif
audio_control_dsp(AUDDRV_DSPCMD_MUTE_DSP_UL, 0, 0, 0, 0, 0);
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_ENABLE, FALSE, 0, 0, 0, 0);
currVoiceMic = AUDIO_SOURCE_UNDEFINED;
}
/*=============================================================================
//
// Function Name: AUDDRV_SetAudioMode
//
// Description: set audio mode for voice call.
//
//=============================================================================
*/
void AUDDRV_SetAudioMode(AudioMode_t audio_mode, AudioApp_t audio_app,
CSL_CAPH_PathID ulPathID,
CSL_CAPH_PathID ul2PathID,
CSL_CAPH_PathID dlPathID)
{
SetAudioMode_Sp_t sp_struct;
aTrace(LOG_AUDIO_DRIVER,
"%s mode==%d, app=%d\n\r", __func__,
audio_mode, audio_app);
#ifdef CONFIG_BCM_MODEM
RPC_SetProperty(RPC_PROP_AUDIO_MODE,
(UInt32) (audio_mode +
audio_app * AUDIO_MODE_NUMBER));
#endif
audio_control_generic(AUDDRV_CPCMD_PassAudioMode,
(UInt32) audio_mode, (UInt32) audio_app, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetAudioMode,
(UInt32) (audio_mode +
audio_app * AUDIO_MODE_NUMBER),
(UInt32) audio_app, 0, 0, 0);
/*load speaker EQ filter and Mic EQ filter from sysparm to DSP*/
/* It means mic1, mic2, speaker */
if (userEQOn == FALSE) {
/* Use the old code, before CP function
audio_control_BuildDSPUlCompfilterCoef() is updated to
handle the mode properly.*/
if (audio_app == AUDIO_APP_VOICE_CALL_WB)
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + AUDIO_MODE_NUMBER, 7, 0, 0, 0);
else
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode % AUDIO_MODE_NUMBER, 7, 0, 0, 0);
/*
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app*AUDIO_MODE_NUMBER,
7, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app * AUDIO_MODE_NUMBER,
1, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app * AUDIO_MODE_NUMBER,
2, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app * AUDIO_MODE_NUMBER,
4, 0, 0, 0);
*/
}
/* else */
/*There is no need for this function to load the ECI-headset-provided
speaker EQ filter and Mic EQ filter to DSP.
//The ECI headset enable/disable request comes with the data.
It means we'll get the coefficients every time if ECI headset on. */
/* audio_cmf_filter((AudioCompfilter_t *) &copy_of_AudioCompfilter ); */
AUDDRV_SetAudioMode_Mic(audio_mode, audio_app, ulPathID, ul2PathID);
sp_struct.mode = audio_mode;
sp_struct.app = audio_app;
if (AUDCTRL_InVoiceCall())
sp_struct.pathID = telephonyPathID.dlPathID;
else
sp_struct.pathID = dlPathID;
sp_struct.inHWlpbk = FALSE;
sp_struct.mixInGain_mB = GAIN_SYSPARM;
sp_struct.mixInGainR_mB = GAIN_SYSPARM;
sp_struct.mixOutGain_mB = GAIN_SYSPARM;
sp_struct.mixOutGainR_mB = GAIN_SYSPARM;
AUDDRV_SetAudioMode_Speaker(sp_struct);
}
/*=============================================================================
//
// Function Name: AUDDRV_SetAudioModeBT
//
// Description: set audio mode for BT voice call. The mode and app is set
// based on the BT device paired with the DUT.
//
//=============================================================================
*/
void AUDDRV_SetAudioModeBT(AudioMode_t audio_mode, AudioApp_t audio_app,
CSL_CAPH_PathID ulPathID,
CSL_CAPH_PathID ul2PathID,
CSL_CAPH_PathID dlPathID)
{
SetAudioMode_Sp_t sp_struct;
aTrace(LOG_AUDIO_DRIVER,
"%s mode==%d, app=%d\n\r", __func__,
audio_mode, audio_app);
#ifdef CONFIG_BCM_MODEM
RPC_SetProperty(RPC_PROP_AUDIO_MODE,
(UInt32) (audio_mode +
audio_app * AUDIO_MODE_NUMBER));
#endif
audio_control_generic(AUDDRV_CPCMD_PassAudioMode,
(UInt32) audio_mode, (UInt32) audio_app, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetAudioBT_Grp,
(UInt32) (audio_mode +
audio_app * AUDIO_MODE_NUMBER),
(UInt32) audio_app, 0, 0, 0);
/*load speaker EQ filter and Mic EQ filter from sysparm to DSP*/
/* It means mic1, mic2, speaker */
if (userEQOn == FALSE) {
/* Use the old code, before CP function
audio_control_BuildDSPUlCompfilterCoef() is updated to
handle the mode properly.*/
if (audio_app == AUDIO_APP_VOICE_CALL_WB)
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + AUDIO_MODE_NUMBER, 7, 0, 0, 0);
else
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode % AUDIO_MODE_NUMBER, 7, 0, 0, 0);
/*
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app*AUDIO_MODE_NUMBER,
7, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app * AUDIO_MODE_NUMBER,
1, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app * AUDIO_MODE_NUMBER,
2, 0, 0, 0);
audio_control_generic(AUDDRV_CPCMD_SetFilter,
audio_mode + audio_app * AUDIO_MODE_NUMBER,
4, 0, 0, 0);
*/
}
/* else */
/*There is no need for this function to load the ECI-headset-provided
speaker EQ filter and Mic EQ filter to DSP.
//The ECI headset enable/disable request comes with the data.
It means we'll get the coefficients every time if ECI headset on. */
/* audio_cmf_filter((AudioCompfilter_t *) &copy_of_AudioCompfilter ); */
AUDDRV_SetAudioMode_Mic(audio_mode, audio_app, ulPathID, ul2PathID);
sp_struct.mode = audio_mode;
sp_struct.app = audio_app;
if (AUDCTRL_InVoiceCall())
sp_struct.pathID = telephonyPathID.dlPathID;
else
sp_struct.pathID = dlPathID;
sp_struct.inHWlpbk = FALSE;
sp_struct.mixInGain_mB = GAIN_SYSPARM;
sp_struct.mixInGainR_mB = GAIN_SYSPARM;
sp_struct.mixOutGain_mB = GAIN_SYSPARM;
sp_struct.mixOutGainR_mB = GAIN_SYSPARM;
AUDDRV_SetAudioMode_Speaker(sp_struct);
}
#ifdef CONFIG_ENABLE_SSMULTICAST
/*=============================================================================
//
// Function Name: AUDDRV_SetAudioMode_Multicast
//
// Description: set audio mode for IHF+HS Multicast.
// For SS multicase both HS and IHF params are from mode SPEAKER
//=============================================================================
*/
void AUDDRV_SetAudioMode_Multicast(SetAudioMode_Sp_t param)
{
int mixInGain, mixOutGain, mixBitSel; /* Register value. */
int mixInGainR, mixOutGainR, mixBitSelR; /* Register value. */
CSL_CAPH_HWConfig_Table_t *path = NULL;
CSL_CAPH_MIXER_e outChnl = CSL_CAPH_SRCM_CH_NONE;
SysMultimediaAudioParm_t *p1;
#if defined(CONFIG_BCM_MODEM) && (!defined(JAVA_ZEBU_TEST))
SysAudioParm_t *p = NULL;
#else
AudioSysParm_t *p = NULL;
#endif
/*For SS multicast case always load params from mode AUDIO_MODE_SPEAKERPHONE*/
if (param.app >= AUDIO_APP_NUMBER) {
p1 = &(MMAudParmP()[AUDIO_MODE_SPEAKERPHONE
+ (param.app - AUDIO_APP_NUMBER) * AUDIO_MODE_NUMBER]);
if (p1 == NULL) {
aError("Multimedia sysparm pointer is NULL\n");
return;
}
} else {
p = &(AudParmP()[AUDIO_MODE_SPEAKERPHONE
+ param.app * AUDIO_MODE_NUMBER]);
if (p == NULL) {
aError("Audio sysparm pointer is NULL\n");
return;
}
}
aTrace(LOG_AUDIO_DRIVER, "%s mode=%d, app %d, pathID %d\n",
__func__, param.mode, param.app, param.pathID);
mixInGain = mixOutGain = mixBitSel = 0;
mixInGainR = mixOutGainR = mixBitSelR = 0;
/* Load the speaker gains form sysparm. */
/*determine which mixer output to apply the gains to */
switch (param.mode) {
case AUDIO_MODE_HEADSET:
case AUDIO_MODE_TTY:
outChnl = CSL_CAPH_SRCM_STEREO_CH1;
break;
case AUDIO_MODE_SPEAKERPHONE:
outChnl = csl_caph_FindMixer(CSL_CAPH_DEV_IHF, 0);
break;
default:
break;
}
/*Load HW Mixer gains from sysparm */
if (param.mixInGain_mB == GAIN_SYSPARM &&
param.mixInGainR_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
mixInGain = (short)p1->srcmixer_input_gain_l;
/* Q13p2 dB */
else
mixInGain = (short)p->srcmixer_input_gain_l;
/* Q13p2 dB */
mixInGain = mixInGain * 25; /* into mB */
if (param.app >= AUDIO_APP_NUMBER)
mixInGainR = (short)p1->srcmixer_input_gain_r;
/* Q13p2 dB */
else
mixInGainR = (short)p->srcmixer_input_gain_r;
/* Q13p2 dB */
mixInGainR = mixInGainR * 25; /* into mB */
} else if (param.mixInGain_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
mixInGain = (short)p1->srcmixer_input_gain_l;
/* Q13p2 dB */
else
mixInGain = (short)p->srcmixer_input_gain_l;
/* Q13p2 dB */
mixInGain = mixInGain * 25; /* into mB */
mixInGainR = param.mixInGainR_mB;
} else if (param.mixInGainR_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
mixInGainR = (short)p1->srcmixer_input_gain_r;
/* Q13p2 dB */
else
mixInGainR = (short)p->srcmixer_input_gain_r;
/* Q13p2 dB */
mixInGainR = mixInGainR * 25; /* into mB */
mixInGain = param.mixInGain_mB;
} else {
mixInGain = param.mixInGain_mB;
mixInGainR = param.mixInGainR_mB;
}
/*determine which which mixer input to apply the gains to */
if (param.pathID >= 1)
path = csl_caph_FindPath(param.pathID);
if (path != 0) {
int i, j;
for (i = 0; i < MAX_SINK_NUM; i++)
for (j = 0; j < MAX_BLOCK_NUM; j++)
if (path->srcmRoute[i][j].outChnl !=
CSL_CAPH_SRCM_CH_NONE) {
outChnl = path->srcmRoute[i][j].outChnl;
aTrace(LOG_AUDIO_DRIVER,
"%s pathID %d outChnl 0x%x inChnl 0x%x\n",
__func__, param.pathID, outChnl,
path->srcmRoute[i][j].inChnl);
aTrace(LOG_AUDIO_DRIVER,
"mixInGain 0x%x, mixInGainR 0x%x\n",
mixInGain, mixInGainR);
csl_srcmixer_setMixInGain(
path->srcmRoute[i][j].inChnl,
path->srcmRoute[i][j].outChnl,
mixInGain, mixInGainR);
}
} else {
aError(
"AUDDRV_SetAudioMode_Speaker can not find path\n");
}
if (outChnl) {
if (param.mixOutGain_mB == GAIN_SYSPARM &&
param.mixOutGainR_mB == GAIN_SYSPARM) {
/* Q13p2 dB */
if (param.app >= AUDIO_APP_NUMBER)
mixOutGain = (short)p1->srcmixer_output_fine_gain_l;
else
mixOutGain = (short)p->srcmixer_output_fine_gain_l;
mixOutGain = mixOutGain * 25; /*into mB */
if (param.app >= AUDIO_APP_NUMBER)
mixOutGainR = (short)p1->srcmixer_output_fine_gain_r;
/* Q13p2 dB */
else
mixOutGainR = (short)p->srcmixer_output_fine_gain_r;
mixOutGainR = mixOutGainR * 25; /*into mB */
} else if (param.mixOutGain_mB == GAIN_SYSPARM) {
/* Q13p2 dB */
if (param.app >= AUDIO_APP_NUMBER)
mixOutGain = (short)p1->srcmixer_output_fine_gain_l;
else
mixOutGain = (short)p->srcmixer_output_fine_gain_l;
mixOutGain = mixOutGain * 25; /*into mB */
mixOutGainR = param.mixOutGainR_mB;
} else if (param.mixOutGainR_mB == GAIN_SYSPARM) {
/* Q13p2 dB */
if (param.app >= AUDIO_APP_NUMBER)
mixOutGainR = (short)p1->srcmixer_output_fine_gain_r;
else
mixOutGainR = (short)p->srcmixer_output_fine_gain_r;
mixOutGainR = mixOutGainR * 25; /*into mB */
mixOutGain = param.mixOutGain_mB;
} else {
mixOutGain = param.mixOutGain_mB;
mixOutGainR = param.mixOutGainR_mB;
}
/* Q13p2 dB */
if (param.app >= AUDIO_APP_NUMBER)
mixBitSel = (short)p1->srcmixer_output_coarse_gain_l;
else
mixBitSel = (short)p->srcmixer_output_coarse_gain_l;
mixBitSel = mixBitSel / 24;
/* bit_shift */
if (param.app >= AUDIO_APP_NUMBER)
mixBitSelR = (short)p1->srcmixer_output_coarse_gain_r;
/* Q13p2 dB */
else
mixBitSelR = (short)p->srcmixer_output_coarse_gain_r;
mixBitSelR = mixBitSelR / 24;
/* bit_shift */
aTrace(LOG_AUDIO_DRIVER,
"%s : mixOutGain 0x%x, mixOutGainR 0x%x, "
"mixBitSel %d, mixBitSelR %d\n",
__func__, mixOutGain, mixOutGainR,
mixBitSel, mixBitSelR);
csl_srcmixer_setMixBitSel(outChnl, mixBitSel, mixBitSelR);
csl_srcmixer_setMixOutGain(outChnl, mixOutGain, mixOutGainR);
}
/* Do not enable/disable sidetone path based on sysparm when in HW loopback */
if ((!param.inHWlpbk) && (p != NULL)) {
void *coef = NULL;
UInt16 gain = 0;
UInt16 enable = 0;
enable = p->hw_sidetone_enable;
if (!enable) {
AUDDRV_HW_DisableSideTone(param.mode);
} else {
AUDDRV_HW_EnableSideTone(param.mode);
coef = (void *)&(p->hw_sidetone_eq[0]);
AUDDRV_HW_SetFilter(AUDDRV_SIDETONE_FILTER, coef);
gain = p->hw_sidetone_gain;
csl_caph_audioh_sidetone_set_gain(gain);
}
}
}
#endif
/*=============================================================================
//
// Function Name: AUDDRV_SetAudioMode_Speaker
//
// Description: set audio mode.
//
//=============================================================================
*/
void AUDDRV_SetAudioMode_Speaker(SetAudioMode_Sp_t param)
{
int mixInGain, mixOutGain, mixBitSel; /* Register value. */
int mixInGainR, mixOutGainR, mixBitSelR; /* Register value. */
int ihf_prot_enable, niir_coeff, gain_attack_step;
int gain_attack_thres, gain_attack_slope, gain_decay_slope;
CSL_CAPH_HWConfig_Table_t *path = NULL;
CSL_CAPH_MIXER_e outChnl = CSL_CAPH_SRCM_CH_NONE;
CSL_CAPH_MIXER_e outChnl1 = CSL_CAPH_SRCM_CH_NONE;
SysMultimediaAudioParm_t *p1;
#if defined(CONFIG_BCM_MODEM) && (!defined(JAVA_ZEBU_TEST))
SysAudioParm_t *p;
#else
AudioSysParm_t *p;
#endif
/* Make sure p is NULL, because later the code will
* access p->hw_sidetone_enable*/
p = NULL;
if (param.app >= AUDIO_APP_NUMBER) {
p1 = &(MMAudParmP()[param.mode
+ (param.app - AUDIO_APP_NUMBER) * AUDIO_MODE_NUMBER]);
if (p1 == NULL) {
aError("Multimedia sysparm pointer is NULL\n");
return;
}
} else {
p = &(AudParmP()[param.mode + param.app * AUDIO_MODE_NUMBER]);
if (p == NULL) {
aError("Audio sysparm pointer is NULL\n");
return;
}
}
aTrace(LOG_AUDIO_DRIVER, "%s mode=%d, app %d, pathID %d\n",
__func__, param.mode, param.app, param.pathID);
mixInGain = mixOutGain = mixBitSel = 0;
mixInGainR = mixOutGainR = mixBitSelR = 0;
ihf_prot_enable = 0;
niir_coeff = 0;
gain_attack_step = 0;
gain_attack_thres = 0;
gain_attack_slope = 0;
gain_decay_slope = 0;
/* Load the speaker gains form sysparm. */
/*determine which mixer output to apply the gains to */
switch (param.mode) {
case AUDIO_MODE_HANDSET:
case AUDIO_MODE_HAC:
outChnl = CSL_CAPH_SRCM_STEREO_CH2_L;
break;
case AUDIO_MODE_HEADSET:
case AUDIO_MODE_TTY:
/*outChnl = (CSL_CAPH_SRCM_STEREO_CH1_L | CSL_CAPH_SRCM_STEREO_CH1_R);*/
outChnl = CSL_CAPH_SRCM_STEREO_CH1;
break;
case AUDIO_MODE_SPEAKERPHONE:
outChnl = csl_caph_FindMixer(CSL_CAPH_DEV_IHF, 0);
break;
case AUDIO_MODE_BLUETOOTH:
#ifdef CONFIG_AUDIO_FEATURE_SET_DISABLE_ECNS
case AUDIO_MODE_HANDSFREE:
#endif
outChnl =
csl_caph_hwctrl_GetMixerOutChannel(CSL_CAPH_DEV_BT_SPKR);
break;
default:
break;
}
/*Load HW Mixer gains from sysparm */
if (param.mixInGain_mB == GAIN_SYSPARM &&
param.mixInGainR_mB == GAIN_SYSPARM) {
/*GAIN_SYSPARM means use sysparm*/
#ifdef JAVA_ZEBU_TEST
if (param.app >= AUDIO_APP_NUMBER)
mixInGain = 0;
#else
if (param.app >= AUDIO_APP_NUMBER)
mixInGain
= (short)p1->srcmixer_input_gain_l; /* Q13p2 dB */
#endif
else
#ifdef JAVA_ZEBU_TEST
mixInGain = 0;
#else
mixInGain
= (short)p->srcmixer_input_gain_l; /* Q13p2 dB */
#endif
mixInGain = mixInGain * 25; /* into mB */
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixInGainR = 0;
#else
mixInGainR
= (short)p1->srcmixer_input_gain_r; /* Q13p2 dB */
#endif
else
#ifdef JAVA_ZEBU_TEST
mixInGainR = 0;
#else
mixInGainR
= (short)p->srcmixer_input_gain_r; /* Q13p2 dB */
#endif
mixInGainR = mixInGainR * 25; /* into mB */
} else if (param.mixInGain_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixInGain = 0;
#else
mixInGain
= (short)p1->srcmixer_input_gain_l; /* Q13p2 dB */
#endif
else
#ifdef JAVA_ZEBU_TEST
mixInGain = 0;
#else
mixInGain
= (short)p->srcmixer_input_gain_l; /* Q13p2 dB */
#endif
mixInGain = mixInGain * 25; /* into mB */
mixInGainR = param.mixInGainR_mB;
} else if (param.mixInGainR_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixInGainR = 0;
#else
mixInGainR
= (short)p1->srcmixer_input_gain_r; /* Q13p2 dB */
#endif
else
#ifdef JAVA_ZEBU_TEST
mixInGainR = 0;
#else
mixInGainR
= (short)p->srcmixer_input_gain_r; /* Q13p2 dB */
#endif
mixInGainR = mixInGainR * 25; /* into mB */
mixInGain = param.mixInGain_mB;
} else {
mixInGain = param.mixInGain_mB;
mixInGainR = param.mixInGainR_mB;
}
/*determine which which mixer input to apply the gains to */
if (param.pathID >= 1)
path = csl_caph_FindPath(param.pathID);
if (path != 0) {
/*if (path->sink[0] == CSL_CAPH_DEV_DSP_throughMEM)
outChnl = path->srcmRoute[0][0].outChnl;*/
int i, j, found;
found = 0;
for (i = 0; i < MAX_SINK_NUM; i++)
for (j = 0; j < MAX_BLOCK_NUM; j++)
if (path->srcmRoute[i][j].outChnl !=
CSL_CAPH_SRCM_CH_NONE) {
/*and supposedly
path->srcmRoute[i][j].sink
matches this speaker*/
outChnl1 = path->srcmRoute[i][j].outChnl;
aTrace(LOG_AUDIO_DRIVER,
"%s pathID %d found outChnl1 0x%x"
" inChnl 0x%x\n",
__func__, param.pathID, outChnl1,
path->srcmRoute[i][j].inChnl);
aTrace(LOG_AUDIO_DRIVER,
"mixInGain 0x%x, mixInGainR 0x%x\n",
mixInGain, mixInGainR);
csl_srcmixer_setMixInGain(
path->srcmRoute[i][j].inChnl,
path->srcmRoute[i][j].outChnl,
mixInGain, mixInGainR);
found = 1;
}
if (found == 0)
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_SetAudioMode_Speaker can not find mixer output\n");
} else {
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_SetAudioMode_Speaker path == NULL, return.\n");
return;
/*do not know which mix input to apply gain on
if (outChnl)
csl_srcmixer_setMixAllInGain(outChnl,
mixInGain, mixInGain);
*/
}
if (outChnl == CSL_CAPH_SRCM_CH_NONE) {
outChnl = outChnl1;
}
if (outChnl) {
if (param.mixOutGain_mB == GAIN_SYSPARM &&
param.mixOutGainR_mB == GAIN_SYSPARM) {
/* Q13p2 dB */
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixOutGain = 0;
#else
mixOutGain
= (short)p1->srcmixer_output_fine_gain_l;
#endif
else
#ifdef JAVA_ZEBU_TEST
mixOutGain = 0;
#else
mixOutGain
= (short)p->srcmixer_output_fine_gain_l;
#endif
mixOutGain = mixOutGain * 25; /*into mB */
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixOutGainR = 0;
#else
mixOutGainR
= (short)p1->srcmixer_output_fine_gain_r;
#endif
else
#ifdef JAVA_ZEBU_TEST
mixOutGainR = 0;
#else
mixOutGainR
= (short)p->srcmixer_output_fine_gain_r;
#endif
mixOutGainR = mixOutGainR * 25; /*into mB */
} else if (param.mixOutGain_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixOutGain = 0;
#else
mixOutGain
= (short)p1->srcmixer_output_fine_gain_l;
#endif
else
#ifdef JAVA_ZEBU_TEST
mixOutGain = 0;
#else
mixOutGain
= (short)p->srcmixer_output_fine_gain_l;
#endif
mixOutGain = mixOutGain * 25; /*into mB */
mixOutGainR = param.mixOutGainR_mB;
} else if (param.mixOutGainR_mB == GAIN_SYSPARM) {
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixOutGainR = 0;
#else
mixOutGainR
= (short)p1->srcmixer_output_fine_gain_r;
#endif
else
#ifdef JAVA_ZEBU_TEST
mixOutGainR = 0;
#else
mixOutGainR
= (short)p->srcmixer_output_fine_gain_r;
#endif
mixOutGainR = mixOutGainR * 25; /*into mB */
mixOutGain = param.mixOutGain_mB;
} else {
mixOutGain = param.mixOutGain_mB;
mixOutGainR = param.mixOutGainR_mB;
}
/* Q13p2 dB */
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixBitSel = 96;
#else
mixBitSel = (short)p1->srcmixer_output_coarse_gain_l;
#endif
else
#ifdef JAVA_ZEBU_TEST
mixBitSel = 96;
#else
mixBitSel = (short)p->srcmixer_output_coarse_gain_l;
#endif
mixBitSel = mixBitSel / 24;
/* bit_shift */
if (param.app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
mixBitSelR = 96;
#else
mixBitSelR = (short)p1->srcmixer_output_coarse_gain_r;
#endif
else
#ifdef JAVA_ZEBU_TEST
mixBitSelR = 96;
#else
mixBitSelR = (short)p->srcmixer_output_coarse_gain_r;
#endif
mixBitSelR = mixBitSelR / 24;
/* bit_shift */
aTrace(LOG_AUDIO_DRIVER,
"%s : mixOutGain 0x%x, mixOutGainR 0x%x, "
"mixBitSel %d, mixBitSelR %d\n",
__func__, mixOutGain, mixOutGainR,
mixBitSel, mixBitSelR);
csl_srcmixer_setMixBitSel(outChnl, mixBitSel, mixBitSelR);
csl_srcmixer_setMixOutGain(outChnl, mixOutGain, mixOutGainR);
/* hardware compressor initialization */
if (param.app >= AUDIO_APP_NUMBER)
ihf_prot_enable = (short)p1->ihf_protection_enable;
if (ihf_prot_enable) {
csl_srcmixer_aldc_control(outChnl, TRUE);
csl_srcmixer_spkrgain_compresser_control(outChnl, TRUE);
niir_coeff = (short)p1->mpm_niir_coeff;
csl_srcmixer_load_spkrgain_iir_coeff(outChnl,
(UInt8 *)&niir_coeff);
gain_attack_step = (short)p1->mpm_gain_attack_step;
gain_attack_thres = (short)p1->mpm_gain_attack_thre;
csl_srcmixer_set_spkrgain_compresser_attack(outChnl,
gain_attack_step, gain_attack_thres);
gain_attack_slope = p1->mpm_gain_attack_slope;
csl_srcmixer_set_spkrgain_compresser_attackslope
(outChnl, gain_attack_slope);
csl_srcmixer_spkrgain_compresser_attackslope_control
(outChnl, TRUE);
gain_decay_slope = p1->mpm_gain_decay_slope;
csl_srcmixer_set_spkrgain_compresser_decayslope
(outChnl, gain_decay_slope);
csl_srcmixer_spkrgain_compresser_decayslope_control
(outChnl, TRUE);
aTrace(LOG_AUDIO_DRIVER,
"%s : ihf_prot_enable %d, niir_coeff 0x%x, "
"gain_attack_step 0x%x, gain_attack_thres 0x%x, "
"gain_attack_slope 0x%x, gain_decay_slope 0x%x\n",
__func__, ihf_prot_enable, niir_coeff,
gain_attack_step, gain_attack_thres,
gain_attack_slope, gain_decay_slope);
}
}
if (path != 0) {
if (path->sink[0] == CSL_CAPH_DEV_DSP_throughMEM)
return;
/*no need for anything other than HW mixer gain for arm2sp */
}
/* Do not enable/disable sidetone path based on sysparm when in HW loopback */
if ((!param.inHWlpbk) && (p != NULL)) {
/*Config sidetone */
void *coef = NULL;
UInt16 gain = 0;
UInt16 enable = 0;
enable = p->hw_sidetone_enable;
if (!enable) {
AUDDRV_HW_DisableSideTone(param.mode);
} else {
/*first step: enable sidetone */
AUDDRV_HW_EnableSideTone(param.mode);
/*second step: set filter and gain. */
coef = (void *)&(p->hw_sidetone_eq[0]);
AUDDRV_HW_SetFilter(AUDDRV_SIDETONE_FILTER, coef);
gain = p->hw_sidetone_gain;
csl_caph_audioh_sidetone_set_gain(gain);
}
}
}
/*=============================================================================
//
// Function Name: AUDDRV_SetAudioMode_Mic
//
// Description: set audio mode.
//
//=============================================================================
*/
void AUDDRV_SetAudioMode_Mic(AudioMode_t audio_mode,
AudioApp_t app,
unsigned int arg_pathID,
unsigned int arg_pathID2)
{
/* for 48 KHz recording no DSP and 8KHz recording through DSP. */
UInt16 gainTemp1 = 0, gainTemp2 = 0, gainTemp3 = 0, gainTemp4 = 0;
SysMultimediaAudioParm_t *p1;
#if defined(CONFIG_BCM_MODEM) && (!defined(JAVA_ZEBU_TEST))
SysAudioParm_t *p;
#else
AudioSysParm_t *p;
#endif
if (app >= AUDIO_APP_NUMBER) {
p1 = &(MMAudParmP()[audio_mode
+ (app - AUDIO_APP_NUMBER) * AUDIO_MODE_NUMBER]);
if (p1 == NULL) {
aError("Multimedia sysparm pointer is NULL\n");
return;
}
} else {
p = &(AudParmP()[audio_mode + app * AUDIO_MODE_NUMBER]);
if (p == NULL) {
aError("Audio sysparm pointer is NULL\n");
return;
}
}
/* Load the mic gains from sysparm. */
/***
do not touch DSP UL gain in this function.
if(isDSPNeeded == TRUE)
{
dspULGain = 64;
//AudParmP()[mode].echoNlp_parms.echo_nlp_gain;
audio_control_generic( AUDDRV_CPCMD_SetBasebandUplinkGain,
dspULGain, 0, 0, 0, 0);
}
***/
if (app >= AUDIO_APP_NUMBER)
#ifdef JAVA_ZEBU_TEST
gainTemp1 = 72;
#else
gainTemp1 = p1->mic_pga; /* Q13p2 */
#endif
else
#ifdef JAVA_ZEBU_TEST
gainTemp1 = 72;
#else
gainTemp1 = p->mic_pga; /* Q13p2 */
#endif
csl_caph_audioh_setMicPga_by_mB(gainTemp1 * 25);
aTrace(LOG_AUDIO_DRIVER,
"%s mode=%d, app=%d mic_gain %d", __func__, audio_mode,
app, gainTemp1*25);
if (app >= AUDIO_APP_NUMBER) {
#ifdef JAVA_ZEBU_TEST
gainTemp1 = 0;
gainTemp2 = 0;
gainTemp1 = 0;
/* dmic1_dga_coarse_gain is the same
* register as amic_dga_coarse_gain */
gainTemp2 = 0;
gainTemp3 = 0;
gainTemp4 = 0;
#else
gainTemp1 = p1->amic_dga_coarse_gain; /* Q13p2 dB */
gainTemp2 = p1->amic_dga_fine_gain; /* Q13p2 dB */
gainTemp1 = p1->dmic1_dga_coarse_gain;
/* dmic1_dga_coarse_gain is the same
* register as amic_dga_coarse_gain */
gainTemp2 = p1->dmic1_dga_fine_gain;
gainTemp3 = p1->dmic2_dga_coarse_gain;
gainTemp4 = p1->dmic2_dga_fine_gain;
#endif
} else {
#ifdef JAVA_ZEBU_TEST
gainTemp1 = 0;
gainTemp2 = 0;
gainTemp1 = 0;
/* dmic1_dga_coarse_gain is the same
* register as amic_dga_coarse_gain */
gainTemp2 = 0;
gainTemp3 = 0;
gainTemp4 = 0;
#else
gainTemp1 = p->amic_dga_coarse_gain; /* Q13p2 dB */
gainTemp2 = p->amic_dga_fine_gain; /* Q13p2 dB */
gainTemp1 = p->dmic1_dga_coarse_gain;
/* dmic1_dga_coarse_gain is the same
* register as amic_dga_coarse_gain */
gainTemp2 = p->dmic1_dga_fine_gain;
gainTemp3 = p->dmic2_dga_coarse_gain;
gainTemp4 = p->dmic2_dga_fine_gain;
#endif
}
csl_caph_audioh_vin_set_cic_scale_by_mB(((int)gainTemp1) * 25,
((int)gainTemp2) * 25,
((int)gainTemp3) * 25,
((int)gainTemp4) * 25);
if (app >= AUDIO_APP_NUMBER) {
#ifdef JAVA_ZEBU_TEST
gainTemp1 = 0;
gainTemp2 = 0;
gainTemp3 = 0;
gainTemp4 = 0;
#else
gainTemp1 = p1->dmic3_dga_coarse_gain;
gainTemp2 = p1->dmic3_dga_fine_gain;
gainTemp3 = p1->dmic4_dga_coarse_gain;
gainTemp4 = p1->dmic4_dga_fine_gain;
#endif
} else {
#ifdef JAVA_ZEBU_TEST
gainTemp1 = 0;
gainTemp2 = 0;
gainTemp3 = 0;
gainTemp4 = 0;
#else
gainTemp1 = p->dmic3_dga_coarse_gain;
gainTemp2 = p->dmic3_dga_fine_gain;
gainTemp3 = p->dmic4_dga_coarse_gain;
gainTemp4 = p->dmic4_dga_fine_gain;
#endif
}
csl_caph_audioh_nvin_set_cic_scale_by_mB(((int)gainTemp1) * 25,
((int)gainTemp2) * 25,
((int)gainTemp3) * 25,
((int)gainTemp4) * 25);
}
/*=============================================================================
//
// Function Name: AUDDRV_User_CtrlDSP
//
// Description: Control DSP Algo
//
//=============================================================================
*/
int AUDDRV_User_CtrlDSP(AudioDrvUserCtrl_t UserCtrlType, Boolean enable,
Int32 size, void *param)
{
#if defined(CONFIG_BCM_MODEM)
static Boolean ConfigSP = FALSE;
static void *spCtrl, *spVar;
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_User_CtrlDSP");
switch (UserCtrlType) {
case AUDDRV_USER_SP_QUERY:
aTrace(LOG_AUDIO_DRIVER, "%s AUDDRV_USER_SP_QUERY", __func__);
if (param == NULL)
return -EINVAL;
csl_dsp_sp_query_msg((UInt32 *) param);
break;
case AUDDRV_USER_SP_CTRL:
aTrace(LOG_AUDIO_DRIVER, "%s AUDDRV_USER_SP_CTRL", __func__);
if (enable) {
if (size <= 0 || param == NULL)
return -EINVAL;
if (spVar != NULL)
csl_dsp_sp_cnfg_msg((UInt16) enable, 0, 1,
(UInt32 *) param, spVar);
else {
spCtrl = kzalloc(size, GFP_KERNEL);
if (!spCtrl)
return -ENOMEM;
memcpy(spCtrl, param, size);
ConfigSP = TRUE;
}
} else {
if (spVar != NULL)
kfree(spVar);
if (spCtrl != NULL)
kfree(spCtrl);
spVar = NULL;
spCtrl = NULL;
csl_dsp_sp_cnfg_msg((UInt16) enable, 0, 0, NULL, NULL);
}
break;
case AUDDRV_USER_SP_VAR:
aTrace(LOG_AUDIO_DRIVER, "%s AUDDRV_USER_SET_SP_VAR",
__func__);
if (ConfigSP == TRUE && spCtrl != NULL) {
if (param == NULL)
return -EINVAL;
csl_dsp_sp_cnfg_msg(1, 0, 1, spCtrl,
(UInt32 *) param);
ConfigSP = FALSE;
} else {
if (size <= 0 || param == NULL)
return -EINVAL;
if (spVar == NULL) {
spVar = kzalloc(size, GFP_KERNEL);
if (!spVar)
return -ENOMEM;
memcpy(spVar, param, size);
}
csl_dsp_sp_ctrl_msg((UInt32 *) param);
}
break;
case AUDDRV_USER_EQ_CTRL:
aTrace(LOG_AUDIO_DRIVER, "%s AUDDRV_USER_EQ_CTRL", __func__);
if (enable == TRUE) {
if (param == NULL)
return -EINVAL;
userEQOn = TRUE;
audio_cmf_filter((AudioCompfilter_t *) param);
} else
userEQOn = FALSE;
break;
default:
aTrace(LOG_AUDIO_DRIVER, "%s Invalid request %d\n\r",
__func__, UserCtrlType);
break;
}
#else
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_User_CtrlDSP:dummy for AP only");
#endif
return 0;
}
/*=============================================================================
//
// Function Name: AUDDRV_User_HandleDSPInt
//
// Description: Handle DSP Interrupt
//
//=============================================================================
*/
void AUDDRV_User_HandleDSPInt(UInt32 param1, UInt32 param2, UInt32 param3)
{
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_User_HandleDSPInt");
if (sUserCB)
((AUDDRV_User_CB) sUserCB) (param1, param2, param3);
}
/*=============================================================================
//
// Function Name: AUDDRV_SetPCMOnOff
//
// Description: set PCM on/off for BT
// this command will be removed from Rhea.
//
//=============================================================================
//=============================================================================
*/
void AUDDRV_ControlFlagFor_CustomGain(Boolean on_off)
{
controlFlagForCustomGain = on_off;
}
/*=============================================================================
//
// Function Name: AUDDRV_SetULSpeechRecordGain
//
// Description: set UL speech recording gain
//
//=============================================================================
*/
void AUDDRV_SetULSpeechRecordGain(Int16 gain)
{
audio_control_generic(AUDDRV_CPCMD_SetULSpeechRecordGain,
(UInt32) gain, 0, 0, 0, 0);
return;
}
/******************************************************************************
*
* Function Name: AUDDRV_SetTuningFlag
*
* Description: set Audio Tuning Flag
*
******************************************************************************/
void AUDDRV_SetTuningFlag(int flag)
{
audio_tuning_flag = flag;
csl_caph_SetTuningFlag(audio_tuning_flag);
}
/******************************************************************************
*
* Function Name: AUDDRV_TuningFlag
*
* Description: read Audio Tuning Flag
*
******************************************************************************/
int AUDDRV_TuningFlag(void)
{
return audio_tuning_flag;
}
/*=============================================================================
// Private function prototypes
//=============================================================================
//=============================================================================
//
// Function Name: AUDDRV_Telephony_InitHW
//
// Description: Enable the HW for Telephone voice call
//
//=============================================================================
*/
static void AUDDRV_Telephony_InitHW(AUDIO_SOURCE_Enum_t mic,
AUDIO_SINK_Enum_t speaker,
AUDIO_SAMPLING_RATE_t sample_rate,
Boolean bNeedDualMic)
{
CSL_CAPH_HWCTRL_CONFIG_t config;
UInt32 *memAddr = 0;
AUDIO_BITS_PER_SAMPLE_t bits = 24;
CSL_CAPH_PathID pathID;
int dl_sink;
AUDIO_SOURCE_Enum_t sec_mic = AUDDRV_GetSecMicFromSpkr(speaker);
#if defined(ENABLE_BT16)
if (speaker == AUDIO_SINK_BTM)
bits = 16;
#endif
aTrace(LOG_AUDIO_DRIVER, "%s mic=%d, spkr=%d sample_rate=%u*\n",
__func__, mic, speaker, sample_rate);
memset(&config, 0, sizeof(CSL_CAPH_HWCTRL_CONFIG_t));
telephonyPathID.ulPathID = 0;
telephonyPathID.ul2PathID = 0;
telephonyPathID.dlPathID = 0;
/* DL */
config.streamID = CSL_CAPH_STREAM_NONE;
config.pathID = 0;
config.secMic = FALSE;
config.source = CSL_CAPH_DEV_DSP;
config.sink = AUDDRV_GetDRVDeviceFromSpkr(speaker);
config.dmaCH = CSL_CAPH_DMA_NONE;
config.src_sampleRate = sample_rate;
config.snk_sampleRate = AUDIO_SAMPLING_RATE_48000;
if (config.sink == CSL_CAPH_DEV_HS)
config.chnlNum = AUDIO_CHANNEL_STEREO;
else
config.chnlNum = AUDIO_CHANNEL_MONO;
config.bitPerSample = bits;
sink = config.sink;
if (sink == CSL_CAPH_DEV_IHF && isIHFDL48k == 1) {
memAddr = AUDIO_GetIHF48KHzBufferBaseAddress();
config.src_sampleRate = AUDIO_SAMPLING_RATE_48000;
config.source = CSL_CAPH_DEV_DSP_throughMEM;
/* csl_caph_EnablePath() handles the case DSP_MEM
when sink is IHF */
csl_caph_hwctrl_setDSPSharedMemForIHF((UInt32) memAddr);
}
telephonyPathID.dlPathID = csl_caph_hwctrl_EnablePath(config);
dl_sink = config.sink;
/* UL */
pathID = csl_caph_FindPathID(CSL_CAPH_DEV_DSP,
AUDDRV_GetDRVDeviceFromMic(mic), 0);
if (pathID) {
/* If voice recording is ongoing, no need to set up UL path.
* Unable to handle this case: record and call use different
* mics.
*/
aTrace(LOG_AUDIO_DRIVER, "%s UL path %d exists\n",
__func__, pathID);
AUDDRV_DisableDSPInput(0);
/*do not set ulPathID before AUDDRV_DisableDSPInput*/
telephonyPathID.ulPathID = pathID;
}
/* Secondary mic */
/* If Dual Mic is enabled. Theoretically DMIC3 or DMIC4 are used*/
if (bNeedDualMic) {
config.streamID = CSL_CAPH_STREAM_NONE;
config.pathID = 0;
config.secMic = TRUE;
config.source = AUDDRV_GetDRVDeviceFromMic(sec_mic);
config.sink = CSL_CAPH_DEV_DSP;
config.dmaCH = CSL_CAPH_DMA_NONE;
config.src_sampleRate = AUDIO_SAMPLING_RATE_48000;
config.snk_sampleRate = sample_rate;
config.chnlNum = AUDIO_CHANNEL_MONO;
config.bitPerSample = bits;
telephonyPathID.ul2PathID = csl_caph_hwctrl_EnablePath(config);
}
/* Primary mic */
config.streamID = CSL_CAPH_STREAM_NONE;
config.pathID = 0;
config.secMic = FALSE;
config.source = AUDDRV_GetDRVDeviceFromMic(mic);
config.sink = CSL_CAPH_DEV_DSP;
config.dmaCH = CSL_CAPH_DMA_NONE;
config.src_sampleRate = AUDIO_SAMPLING_RATE_48000;
config.snk_sampleRate = sample_rate;
config.chnlNum = AUDIO_CHANNEL_MONO;
config.bitPerSample = bits;
if (voiceRecOn && voiceInMic != mic)
aError("%s voice record (%d) and call (%d) different mics\n",
__func__, voiceInMic, mic);
if (!pathID)
telephonyPathID.ulPathID = csl_caph_hwctrl_EnablePath(config);
/*to avoid the pop of the first ep call, start ep rampup last*/
if (dl_sink == CSL_CAPH_DEV_EP)
csl_caph_audioh_start_ep();
return;
}
/*=============================================================================
//
// Function Name: AUDDRV_Telephony_DeinitHW
//
// Description: Disable the HW for Telephone voice call
//
//=============================================================================
*/
void AUDDRV_Telephony_DeinitHW(void)
{
CSL_CAPH_HWCTRL_CONFIG_t config;
aTrace(LOG_AUDIO_DRIVER, "AUDDRV_Telephony_DeinitHW\n");
memset(&config, 0, sizeof(CSL_CAPH_HWCTRL_CONFIG_t));
AUDDRV_HW_DisableSideTone(AUDCTRL_GetAudioMode());
config.streamID = CSL_CAPH_STREAM_NONE;
config.pathID = telephonyPathID.ulPathID;
if (!voiceRecOn)
csl_caph_hwctrl_DisablePath(config);
if (telephonyPathID.ul2PathID != 0) {
config.streamID = CSL_CAPH_STREAM_NONE;
config.pathID = telephonyPathID.ul2PathID;
(void)csl_caph_hwctrl_DisablePath(config);
}
config.streamID = CSL_CAPH_STREAM_NONE;
config.pathID = telephonyPathID.dlPathID;
(void)csl_caph_hwctrl_DisablePath(config);
sink = CSL_CAPH_DEV_NONE;
telephonyPathID.ulPathID = 0;
telephonyPathID.ul2PathID = 0;
telephonyPathID.dlPathID = 0;
return;
}
/****************************************************************************
*
* Function Name: AUDDRV_HW_SetFilter(AUDDRV_HWCTRL_FILTER_e filter,
* void* coeff)
*
* Description: Load filter coefficients
*
****************************************************************************/
static void AUDDRV_HW_SetFilter(AUDDRV_HWCTRL_FILTER_e filter,
void *coeff)
{
if (filter == AUDDRV_SIDETONE_FILTER)
csl_caph_audioh_sidetone_load_filter((UInt32 *) coeff);
}
/****************************************************************************
*
* Function Name: AUDDRV_HW_EnableSideTone(AudioMode_t audio_mode)
*
* Description: Enable Sidetone path
*
****************************************************************************/
static void AUDDRV_HW_EnableSideTone(AudioMode_t audio_mode)
{
AUDDRV_PATH_Enum_t pathId = AUDDRV_PATH_VIBRA_OUTPUT;
switch (audio_mode) {
case AUDIO_MODE_HEADSET:
case AUDIO_MODE_TTY:
pathId = AUDDRV_PATH_HEADSET_OUTPUT;
break;
case AUDIO_MODE_HANDSET:
case AUDIO_MODE_HAC:
pathId = AUDDRV_PATH_EARPICEC_OUTPUT;
break;
case AUDIO_MODE_SPEAKERPHONE:
pathId = AUDDRV_PATH_IHF_OUTPUT;
break;
default:
;
}
csl_caph_audioh_sidetone_control((int)pathId, TRUE);
}
/****************************************************************************
*
* Function Name: AUDDRV_HW_DisableSideTone(AudioMode_t audio_mode)
*
* Description: Disable Sidetone path
*
****************************************************************************/
static void AUDDRV_HW_DisableSideTone(AudioMode_t audio_mode)
{
AUDDRV_PATH_Enum_t pathId = AUDDRV_PATH_VIBRA_OUTPUT;
switch (audio_mode) {
case AUDIO_MODE_HEADSET:
case AUDIO_MODE_TTY:
pathId = AUDDRV_PATH_HEADSET_OUTPUT;
break;
case AUDIO_MODE_HANDSET:
case AUDIO_MODE_HAC:
pathId = AUDDRV_PATH_EARPICEC_OUTPUT;
break;
case AUDIO_MODE_SPEAKERPHONE:
pathId = AUDDRV_PATH_IHF_OUTPUT;
break;
default:
;
}
csl_caph_audioh_sidetone_control((int)pathId, FALSE);
}
/*==========================================================================
//
// Function Name: AUDDRV_GetDRVDeviceFromMic
//
// Description: Get the audio driver Device from the Microphone selection.
//
// =========================================================================
*/
CSL_CAPH_DEVICE_e AUDDRV_GetDRVDeviceFromMic(AUDIO_SOURCE_Enum_t mic)
{
CSL_CAPH_DEVICE_e dev = MIC_Mapping_Table[mic].dev;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_GetDRVDeviceFromMic::mic %d:%d\n", mic, dev);
return dev;
}
/*==========================================================================
//
// Function Name: AUDDRV_GetDRVDeviceFromSpkr
//
// Description: Get the audio driver Device from the Speaker selection.
//
// =========================================================================
*/
CSL_CAPH_DEVICE_e AUDDRV_GetDRVDeviceFromSpkr(AUDIO_SINK_Enum_t spkr)
{
CSL_CAPH_DEVICE_e dev = SPKR_Mapping_Table[spkr].dev;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_GetDRVDeviceFromSpkr::spkr %d:%d\n", spkr, dev);
return dev;
}
/*==========================================================================
//
// Function Name: AUDDRV_GetPrimaryMicFromSpkr
//
// Description: Get the primary mic based on the
// speaker selection.
//
// =========================================================================
*/
AUDIO_SOURCE_Enum_t AUDDRV_GetPrimaryMicFromSpkr(AUDIO_SINK_Enum_t spkr)
{
AUDIO_SOURCE_Enum_t pri_mic = SPKR_MIC_Mapping_Table[spkr].pri_mic;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_GetPrimaryMicFromSpkr::spkr=%d, "
"pri mic=%d\n", spkr, pri_mic);
return pri_mic;
}
/*==========================================================================
//
// Function Name: AUDDRV_GetSecMicFromSpkr
//
// Description: Get the secondary MIC from the speaker selection.
//
// =========================================================================
*/
AUDIO_SOURCE_Enum_t AUDDRV_GetSecMicFromSpkr(AUDIO_SINK_Enum_t spkr)
{
AUDIO_SOURCE_Enum_t sec_mic = SPKR_MIC_Mapping_Table[spkr].sec_mic;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_GetSecMicFromSpkr::spkr=%d, "
"sec mic=%d\n", spkr, sec_mic);
return sec_mic;
}
/*==========================================================================
//
// Function Name: AUDDRV_SetPrimaryMicFromSpkr
//
// Description: Set the primary mic based on the
// speaker selection.
//
// =========================================================================
*/
void AUDDRV_SetPrimaryMicFromSpkr(AUDIO_SINK_Enum_t spkr,
AUDIO_SOURCE_Enum_t mic)
{
SPKR_MIC_Mapping_Table[spkr].pri_mic = mic;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_SetPrimaryMicFromSpkr::spkr=%d, pri mic=%d\n",
spkr, mic);
}
/*==========================================================================
//
// Function Name: AUDDRV_SetSecMicFromSpkr
//
// Description: Set the secondary mic based on the
// speaker selection.
//
// =========================================================================
*/
void AUDDRV_SetSecMicFromSpkr(AUDIO_SINK_Enum_t spkr,
AUDIO_SOURCE_Enum_t mic)
{
SPKR_MIC_Mapping_Table[spkr].sec_mic = mic;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_SetSecMicFromSpkr::spkr=%d, sec mic=%d\n", spkr, mic);
}
/*==========================================================================
// Function Name: AUDDRV_PrintAllMics
//
// Description: Print all primary and secondary mic selection for all sinks
//
// =========================================================================
*/
void AUDDRV_PrintAllMics(void)
{
int i;
aTrace(LOG_AUDIO_DRIVER,
"AUDDRV_PrintAllMics:: print primary and secondary"
" mics for each sink\n");
for (i = 0; i < AUDIO_SINK_VALID_TOTAL; i++) {
aTrace(LOG_AUDIO_DRIVER,
"sink %d:: pri mic=%d, sec mic=%d\n",
i,
SPKR_MIC_Mapping_Table[i].pri_mic,
SPKR_MIC_Mapping_Table[i].sec_mic);
}
}
static UInt32 *AUDIO_GetIHF48KHzBufferBaseAddress(void)
{
/* special path for IHF voice call
need to use the physical address
// Linux only change
AP_SharedMem_t *ap_shared_mem_ptr =
ioremap_nocache(AP_SH_BASE, AP_SH_SIZE);
// Linux only : to get the physical address use the virtual
address to compute offset and
// add to the base address
UInt32 *memAddr = (UInt32 *)(AP_SH_BASE +
((UInt32)&(ap_shared_mem_ptr->shared_aud_out_buf_48k[0][0]) -
(UInt32)ap_shared_mem_ptr));
return memAddr;
*/
#ifdef CONFIG_BCM_MODEM
return AUDIO_Return_IHF_48kHz_buffer_base_address();
#else
UInt32 *memAddr = 0;
return memAddr;
#endif
}
#ifdef CONFIG_BCM_MODEM
static void AP_ProcessAudioEnableDone(UInt16 enabled_path)
{
aTrace(LOG_AUDIO_DRIVER,
"AP_ProcessAudioEnableDone, Got AUDIO ENABLE RESP FROM DSP 0x%x\n",
enabled_path);
#ifdef CONFIG_BCM_MODEM
csl_caph_dspcb(enabled_path & ~((UInt16)DSP_AADMAC_RETIRE_DS_CMD));
#endif
/*aError("i_e");*/
complete(&audioEnableDone);
/*aError("i_f");*/
}
static void AP_ProcessArm2spHqDLInitDone()
{
aTrace(LOG_AUDIO_DRIVER,
"%s, Got Arm2spHqDL RESP FROM DSP\n", __func__);
/*aError("i_e");*/
complete(&arm2spHqDLInitDone);
/*aError("i_f");*/
}
#endif
/****************************************************************************
*
* Function Name: AUDDRV_GetULPath
*
* Description: Get UL path
*
*****************************************************************************/
int AUDDRV_GetULPath(void)
{
int ret = (int)telephonyPathID.ulPathID;
aTrace(LOG_AUDIO_DRIVER, "%s %d\n", __func__, ret);
return ret;
}
void AUDDRV_SetCallMode(Int32 callMode)
{
curCallMode = callMode;
aTrace(LOG_AUDIO_DRIVER, "%s callMode = %d\n",
__func__, (int)curCallMode);
}
void AUDDRV_ConnectDL(void)
{
aTrace(LOG_AUDIO_DRIVER, "%s PTT CONNECT_DL\n", __func__);
#if defined(ENABLE_DMA_VOICE)
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, TRUE, 0, 0, 0, 0);
#else
audio_control_dsp(AUDDRV_DSPCMD_AUDIO_CONNECT_DL, TRUE,
AUDCTRL_Telephony_HW_16K(AUDCTRL_GetAudioMode()), 0, 0, 0);
#endif
}
void AUDDRV_CPResetCleanup(void)
{
currVoiceMic = AUDIO_SOURCE_UNDEFINED;
currVoiceSpkr = AUDIO_SINK_UNDEFINED;
voiceRecOn = FALSE;
voicePlayOutpathEnabled = FALSE;
dspECEnable = TRUE;
dspNSEnable = TRUE;
inCallRateChange = FALSE;
sink = CSL_CAPH_DEV_NONE;
telephonyPathID.ulPathID = 0;
telephonyPathID.ul2PathID = 0;
telephonyPathID.dlPathID = 0;
voiceInMic = AUDIO_SOURCE_UNDEFINED;
voiceInSr = 0;
curCallMode = CALL_MODE_NONE;
init_completion(&audioEnableDone);
}
void AUDDRV_SetCallSampleRate(UInt32 voiceSampleRate)
{
curCallSampleRate = voiceSampleRate;
aTrace(LOG_AUDIO_DRIVER, "%s curCallSampleRate = %d\n",
__func__, (int)curCallSampleRate);
}
void AUDDRV_SetIHFDLSampleRate(int mode)
{
aTrace(LOG_AUDIO_DRIVER, "%s isIHFDL48k %d->%d\n",
__func__, isIHFDL48k, mode);
isIHFDL48k = mode;
}
int AUDDRV_Get_TrEqParm(void *param, unsigned int size, AudioApp_t app,
unsigned int sample_rate)
{
#ifndef CONFIG_BCM_MODEM
return -EINVAL;
#else
SysIndMultimediaAudioParm_t *p;
if (app != AUDIO_APP_MUSIC)
return -EINVAL;
/* Sample rate added to input param */
if (size > (sizeof(*p) + sizeof(unsigned int))) {
aError("SysIndMultimediaAudioParm_t changed %s",
__func__);
BUG();
return -EINVAL;
}
p = APSYSPARM_GetIndMultimediaAudioParmAccessPtr();
if (p == NULL) {
aError("%s cannot read TREQ param", __func__);
return -EINVAL;
}
/* 44.1 uses IHF mode, 48 uses AUDIO_MODE_RESERVE */
if ((AUDIO_SAMPLING_RATE_t)sample_rate
== AUDIO_SAMPLING_RATE_44100)
memcpy(param, p+AUDIO_MODE_SPEAKERPHONE,
(size - sizeof(unsigned int)));
else if ((AUDIO_SAMPLING_RATE_t)sample_rate
== AUDIO_SAMPLING_RATE_48000)
memcpy(param, p+AUDIO_MODE_RESERVE,
(size - sizeof(unsigned int)));
return 0;
#endif
}
/*==========================================================================
//
// Function Name: AUDDRV_SetEchoRefMic
//
// Description: Set the dsp echo ref mic
//
// =========================================================================
*/
void AUDDRV_SetEchoRefMic(int arg1)
{
if (arg1 < 0 || arg1 > 1)
return;
aTrace(LOG_AUDIO_DRIVER,
"%s::old ref_mic =%d, "
"new ref_mic=%d\n", __func__, echo_ref_mic, arg1);
echo_ref_mic = arg1;
}
/*==========================================================================
//
// Function Name: AUDDRV_GetEchoRefMic
//
// Description: Get the dsp echo ref mic
//
// =========================================================================
*/
int AUDDRV_GetEchoRefMic(void)
{
aTrace(LOG_AUDIO_DRIVER,
"%s::echo_ref_mic=%d\n", __func__, echo_ref_mic);
return echo_ref_mic;
}
/*==========================================================================
//
// Function Name: AUDDRV_Get_FDMBCParm
//
// Description: Get the FDMBC tuning params from CP
//
// =========================================================================
*/
int AUDDRV_Get_FDMBCParm(void *param, int size)
{
#ifndef CONFIG_BCM_MODEM
return -EINVAL;
#else
#ifdef CONFIG_ARCH_JAVA
SysIndMultimediaAudioParm_t *p;
if (param == NULL)
return -EINVAL;
p = APSYSPARM_GetIndMultimediaAudioParmAccessPtr();
if (p == NULL) {
aError("%s cannot read MBC param", __func__);
return -EINVAL;
}
memcpy(param, &((p+AUDIO_MODE_SPEAKERPHONE)->mbc_cr[0]), size);
#endif
return 0;
#endif
}