hal/msm8974/platform.h - platform/hardware/qcom/audio - Git at Google

 /*
  * Copyright (C) 2013-2014 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

 /*
  * 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_SPEAKER_SAFE | \
      AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)

 /* 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_SPEAKER_SAFE,
     SND_DEVICE_OUT_HEADPHONES,
     SND_DEVICE_OUT_LINE,
     SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
     SND_DEVICE_OUT_VOICE_HANDSET,
     SND_DEVICE_OUT_VOICE_SPEAKER,
     SND_DEVICE_OUT_VOICE_HEADPHONES,
     SND_DEVICE_OUT_VOICE_LINE,
     SND_DEVICE_OUT_HDMI,
     SND_DEVICE_OUT_SPEAKER_AND_HDMI,
     SND_DEVICE_OUT_BT_SCO,
     SND_DEVICE_OUT_BT_SCO_WB,
     SND_DEVICE_OUT_VOICE_HANDSET_TMUS,
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
     SND_DEVICE_OUT_VOICE_HAC_HANDSET,
     SND_DEVICE_OUT_VOICE_TX,
     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_HANDSET_DMIC_STEREO,

     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_SPEAKER_DMIC_STEREO,

     SND_DEVICE_IN_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_DMIC_TMUS,
     SND_DEVICE_IN_VOICE_SPEAKER_MIC,
     SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
     SND_DEVICE_IN_VOICE_HEADSET_MIC,
     SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,
     SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,
     SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,

     SND_DEVICE_IN_VOICE_REC_MIC,
     SND_DEVICE_IN_VOICE_REC_MIC_NS,
     SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
     SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,

     SND_DEVICE_IN_VOICE_RX,

     SND_DEVICE_IN_END,

     SND_DEVICE_MAX = SND_DEVICE_IN_END,

 };

 #define DEFAULT_OUTPUT_SAMPLING_RATE 48000

 #define ALL_SESSION_VSID                0xFFFFFFFF
 #define DEFAULT_MUTE_RAMP_DURATION_MS   20
 #define DEFAULT_VOLUME_RAMP_DURATION_MS 20

 #ifdef PLATFORM_MSM8084
 #define ACDB_ID_VOICE_SPEAKER 66
 #define ACDB_ID_VOICE_HANDSET 67
 #define ACDB_ID_VOICE_HANDSET_TMUS 67
 #define ACDB_ID_VOICE_DMIC_EF_TMUS 89
 #else
 #define ACDB_ID_VOICE_SPEAKER 15
 #define ACDB_ID_VOICE_HANDSET 7
 #define ACDB_ID_VOICE_HANDSET_TMUS 88
 #define ACDB_ID_VOICE_DMIC_EF_TMUS 89
 #endif

 #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

 #define HDMI_MULTI_PERIOD_SIZE  336
 #define HDMI_MULTI_PERIOD_COUNT 8
 #define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
 #define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)

 #define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
 #define AUDIO_CAPTURE_PERIOD_COUNT 2

 #define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
 #define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
 #define LOW_LATENCY_CAPTURE_USE_CASE 1

 #define DEEP_BUFFER_PCM_DEVICE 0
 #define AUDIO_RECORD_PCM_DEVICE 0
 #define MULTIMEDIA2_PCM_DEVICE 1
 #define PLAYBACK_OFFLOAD_DEVICE 9
 #define LOWLATENCY_PCM_DEVICE 15
 #define VOICE_VSID  0x10C01000
 #ifdef PLATFORM_MSM8084
 #define VOICE_CALL_PCM_DEVICE 20
 #define VOICE2_CALL_PCM_DEVICE 25
 #define VOLTE_CALL_PCM_DEVICE 21
 #define QCHAT_CALL_PCM_DEVICE 33
 #define VOWLAN_CALL_PCM_DEVICE -1
 #else
 #define VOICE_CALL_PCM_DEVICE 2
 #define VOICE2_CALL_PCM_DEVICE 22
 #define VOLTE_CALL_PCM_DEVICE 14
 #define QCHAT_CALL_PCM_DEVICE 20
 #define VOWLAN_CALL_PCM_DEVICE 36
 #endif

 #define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
 #define AFE_PROXY_RECORD_PCM_DEVICE 8

 #define HFP_PCM_RX 5
 #ifdef PLATFORM_MSM8x26
 #define HFP_SCO_RX 28
 #define HFP_ASM_RX_TX 29
 #else
 #define HFP_SCO_RX 23
 #define HFP_ASM_RX_TX 24
 #endif

 #define LIB_CSD_CLIENT "libcsd-client.so"
 #define LIB_MDM_DETECT "libmdmdetect.so"

 /* CSD-CLIENT related functions */
 typedef int (*init_t)(bool);
 typedef int (*deinit_t)();
 typedef int (*disable_device_t)();
 typedef int (*enable_device_config_t)(int, int);
 typedef int (*enable_device_t)(int, int, uint32_t);
 typedef int (*volume_t)(uint32_t, int, uint16_t);
 typedef int (*mic_mute_t)(uint32_t, int, uint16_t);
 typedef int (*slow_talk_t)(uint32_t, uint8_t);
 typedef int (*start_voice_t)(uint32_t);
 typedef int (*stop_voice_t)(uint32_t);
 typedef int (*start_playback_t)(uint32_t);
 typedef int (*stop_playback_t)(uint32_t);
 typedef int (*start_record_t)(uint32_t, int);
 typedef int (*stop_record_t)(uint32_t);
 typedef int (*get_sample_rate_t)(uint32_t *);
 /* CSD Client structure */
 struct csd_data {
     void *csd_client;
     init_t init;
     deinit_t deinit;
     disable_device_t disable_device;
     enable_device_config_t enable_device_config;
     enable_device_t enable_device;
     volume_t volume;
     mic_mute_t mic_mute;
     slow_talk_t slow_talk;
     start_voice_t start_voice;
     stop_voice_t stop_voice;
     start_playback_t start_playback;
     stop_playback_t stop_playback;
     start_record_t start_record;
     stop_record_t stop_record;
     get_sample_rate_t get_sample_rate;
 };

 #endif // QCOM_AUDIO_PLATFORM_H
	/*
	* Copyright (C) 2013-2014 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

	/*
	* 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_SPEAKER_SAFE \| \
	AUDIO_DEVICE_OUT_WIRED_HEADSET \| AUDIO_DEVICE_OUT_WIRED_HEADPHONE)

	/* 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_SPEAKER_SAFE,
	SND_DEVICE_OUT_HEADPHONES,
	SND_DEVICE_OUT_LINE,
	SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
	SND_DEVICE_OUT_VOICE_HANDSET,
	SND_DEVICE_OUT_VOICE_SPEAKER,
	SND_DEVICE_OUT_VOICE_HEADPHONES,
	SND_DEVICE_OUT_VOICE_LINE,
	SND_DEVICE_OUT_HDMI,
	SND_DEVICE_OUT_SPEAKER_AND_HDMI,
	SND_DEVICE_OUT_BT_SCO,
	SND_DEVICE_OUT_BT_SCO_WB,
	SND_DEVICE_OUT_VOICE_HANDSET_TMUS,
	SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
	SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
	SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
	SND_DEVICE_OUT_VOICE_HAC_HANDSET,
	SND_DEVICE_OUT_VOICE_TX,
	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_HANDSET_DMIC_STEREO,

	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_SPEAKER_DMIC_STEREO,

	SND_DEVICE_IN_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_DMIC_TMUS,
	SND_DEVICE_IN_VOICE_SPEAKER_MIC,
	SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
	SND_DEVICE_IN_VOICE_HEADSET_MIC,
	SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,
	SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,
	SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,

	SND_DEVICE_IN_VOICE_REC_MIC,
	SND_DEVICE_IN_VOICE_REC_MIC_NS,
	SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
	SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,

	SND_DEVICE_IN_VOICE_RX,

	SND_DEVICE_IN_END,

	SND_DEVICE_MAX = SND_DEVICE_IN_END,

	};

	#define DEFAULT_OUTPUT_SAMPLING_RATE 48000

	#define ALL_SESSION_VSID 0xFFFFFFFF
	#define DEFAULT_MUTE_RAMP_DURATION_MS 20
	#define DEFAULT_VOLUME_RAMP_DURATION_MS 20

	#ifdef PLATFORM_MSM8084
	#define ACDB_ID_VOICE_SPEAKER 66
	#define ACDB_ID_VOICE_HANDSET 67
	#define ACDB_ID_VOICE_HANDSET_TMUS 67
	#define ACDB_ID_VOICE_DMIC_EF_TMUS 89
	#else
	#define ACDB_ID_VOICE_SPEAKER 15
	#define ACDB_ID_VOICE_HANDSET 7
	#define ACDB_ID_VOICE_HANDSET_TMUS 88
	#define ACDB_ID_VOICE_DMIC_EF_TMUS 89
	#endif

	#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

	#define HDMI_MULTI_PERIOD_SIZE 336
	#define HDMI_MULTI_PERIOD_COUNT 8
	#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
	#define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)

	#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
	#define AUDIO_CAPTURE_PERIOD_COUNT 2

	#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
	#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
	#define LOW_LATENCY_CAPTURE_USE_CASE 1

	#define DEEP_BUFFER_PCM_DEVICE 0
	#define AUDIO_RECORD_PCM_DEVICE 0
	#define MULTIMEDIA2_PCM_DEVICE 1
	#define PLAYBACK_OFFLOAD_DEVICE 9
	#define LOWLATENCY_PCM_DEVICE 15
	#define VOICE_VSID 0x10C01000
	#ifdef PLATFORM_MSM8084
	#define VOICE_CALL_PCM_DEVICE 20
	#define VOICE2_CALL_PCM_DEVICE 25
	#define VOLTE_CALL_PCM_DEVICE 21
	#define QCHAT_CALL_PCM_DEVICE 33
	#define VOWLAN_CALL_PCM_DEVICE -1
	#else
	#define VOICE_CALL_PCM_DEVICE 2
	#define VOICE2_CALL_PCM_DEVICE 22
	#define VOLTE_CALL_PCM_DEVICE 14
	#define QCHAT_CALL_PCM_DEVICE 20
	#define VOWLAN_CALL_PCM_DEVICE 36
	#endif

	#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
	#define AFE_PROXY_RECORD_PCM_DEVICE 8

	#define HFP_PCM_RX 5
	#ifdef PLATFORM_MSM8x26
	#define HFP_SCO_RX 28
	#define HFP_ASM_RX_TX 29
	#else
	#define HFP_SCO_RX 23
	#define HFP_ASM_RX_TX 24
	#endif

	#define LIB_CSD_CLIENT "libcsd-client.so"
	#define LIB_MDM_DETECT "libmdmdetect.so"

	/* CSD-CLIENT related functions */
	typedef int (*init_t)(bool);
	typedef int (*deinit_t)();
	typedef int (*disable_device_t)();
	typedef int (*enable_device_config_t)(int, int);
	typedef int (*enable_device_t)(int, int, uint32_t);
	typedef int (*volume_t)(uint32_t, int, uint16_t);
	typedef int (*mic_mute_t)(uint32_t, int, uint16_t);
	typedef int (*slow_talk_t)(uint32_t, uint8_t);
	typedef int (*start_voice_t)(uint32_t);
	typedef int (*stop_voice_t)(uint32_t);
	typedef int (*start_playback_t)(uint32_t);
	typedef int (*stop_playback_t)(uint32_t);
	typedef int (*start_record_t)(uint32_t, int);
	typedef int (*stop_record_t)(uint32_t);
	typedef int (get_sample_rate_t)(uint32_t );
	/* CSD Client structure */
	struct csd_data {
	void *csd_client;
	init_t init;
	deinit_t deinit;
	disable_device_t disable_device;
	enable_device_config_t enable_device_config;
	enable_device_t enable_device;
	volume_t volume;
	mic_mute_t mic_mute;
	slow_talk_t slow_talk;
	start_voice_t start_voice;
	stop_voice_t stop_voice;
	start_playback_t start_playback;
	stop_playback_t stop_playback;
	start_record_t start_record;
	stop_record_t stop_record;
	get_sample_rate_t get_sample_rate;
	};

	#endif // QCOM_AUDIO_PLATFORM_H