【耳机出来的声音很小，测试的时候请注意将耳机的声音调节为最大！】
1、增加nau8822的配置：
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\codec_config\config_list.h
2、【感觉拿掉这个声卡！^_】
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\codec_config\es8388_config.h
3、【新建nau8822的头问阿金】
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\codec_config\nau8822_config.h
4、【相关修改】
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\audio_hw.c
20250803让飞凌OK3576-C开发板在Rockchip的原厂Android14下适配声卡NAU88C22YG【Android部分】
2025/8/3 19:10
缘起：很容易知道 飞凌OK3576-C开发板 使用的声卡芯片是 NAU88C22YG
新唐科技(NUVOTON) NAU8822LYG NAU88C22YG 新唐立体声音频编解码芯片
原理图：OK3576-C V1.2_20240830.pdf
搜索：HP，耳机，很容易找到第16页/总共21页，
NAU88C22YG
IIC ADDRESS：0011010R/W
0x1A
1、
Y:\OK3576-android-source\hardware\rockchip\audio\tinyalsa_hal\codec_config\config_list.h
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\codec_config\config_list.h
/*
* Copyright (C) 2015 Rockchip Electronics Co., Ltd.
*/
/**
* @file config_list.h
* @brief
* @author RkAudio
* @version 1.0.8
* @date 2015-08-24
*/
#ifndef _CONFIG_LIST_H_
#define _CONFIG_LIST_H_
#include "config.h"
#include "default_config.h"
#include "rk616_config.h"
#include "rt3261_config.h"
#include "rt5616_config.h"
#include "rt5640_config.h"
#include "rt5678_config.h"
#include "rt3224_config.h"
#include "wm8960_config.h"
#include "es8388_config.h"
#include "es8396_config.h"
#include "rt5651_config.h"
#include "es8316_config.h"
#include "cx2072_config.h"
#include "nau8822_config.h"
#include "hdmi_analog_config.h"
struct alsa_sound_card_config
{
const char *sound_card_name;
const struct config_route_table *route_table;
};
/*
* List of sound card name and config table.
* Audio will get config_route_table and set route
* according to the name of sound card 0 and sound_card_name.
*/
struct alsa_sound_card_config sound_card_config_list[] = {
{
.sound_card_name = "RKRK616",
.route_table = &rk616_config_table,
},
{
.sound_card_name = "RK29RT3224",
.route_table = &rt3224_config_table,
},
{
.sound_card_name = "rockchiprt5640c",
.route_table = &rt5640_config_table,
},
{
.sound_card_name = "realtekrt5678co",
.route_table = &rt5678_config_table,
},
{
.sound_card_name = "rockchipes8388",
.route_table = &es8388_config_table,
},
{
.sound_card_name = "rockchipes8388c",
.route_table = &es8388_config_table,
},
{
.sound_card_name = "rockchipes8396c",
.route_table = &es8396_config_table,
},
{
.sound_card_name = "rockchipnau8822",
.route_table = &nau8822_config_table,
},
{
.sound_card_name = "RK29RT3261",
.route_table = &rt3261_config_table,
},
{
.sound_card_name = "RK29WM8960",
.route_table = &wm8960_config_table,
},
{
.sound_card_name = "RKRT3224",
.route_table = &rt3224_config_table,
},
{
.sound_card_name = "RKRT3261",
.route_table = &rt3261_config_table,
},
{
.sound_card_name = "RKWM8960",
.route_table = &wm8960_config_table,
},
{
.sound_card_name = "RKRT5616",
.route_table = &rt5616_config_table,
},
{
.sound_card_name = "realtekrt5651co",
.route_table = &rt5651_config_table,
},
{
.sound_card_name = "rockchipes8316c",
.route_table = &es8316_config_table,
},
{
.sound_card_name = "rockchipcx2072x",
.route_table = &cx2072_config_table,
},
{
.sound_card_name = "rkhdmianalogsnd",
.route_table = &hdmi_analog_config_table,
}
};
#endif //_CONFIG_LIST_H_
2、
Y:\OK3576-android-source\hardware\rockchip\audio\tinyalsa_hal\codec_config\es8388_config.h
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\codec_config\es8388_config.h
const struct config_control es8388_main_mic_capture_controls[] = {
{
.ctl_name = "Capture Digital Volume",
.int_val = {192, 192},
},
{
.ctl_name = "Capture Mute",
.int_val = {off},
},
{
.ctl_name = "Left Channel Capture Volume",
.int_val = {8},
},
{
.ctl_name = "Right Channel Capture Volume",
.int_val = {8},
},
{
.ctl_name = "Right PGA Mux",
.str_val = "DifferentialR",
},
{
.ctl_name = "Left PGA Mux",
.str_val = "DifferentialL",
},
{
.ctl_name = "Differential Mux",
.str_val = "Line 2",
},
};
const struct config_control es8388_hands_free_mic_capture_controls[] = {
{
.ctl_name = "Capture Digital Volume",
.int_val = {192, 192},
},
{
.ctl_name = "Capture Mute",
.int_val = {off},
},
{
.ctl_name = "Left Channel Capture Volume",
.int_val = {8},
},
{
.ctl_name = "Right Channel Capture Volume",
.int_val = {8},
},
{
.ctl_name = "Right PGA Mux",
.str_val = "DifferentialR",
},
{
.ctl_name = "Left PGA Mux",
.str_val = "DifferentialL",
},
{
.ctl_name = "Differential Mux",
.str_val = "Line 1",
},
};
修改为：
const struct config_control es8388_main_mic_capture_controls[] = {
{
.ctl_name = "Capture Digital Volume",
.int_val = {192, 192},
},
{
.ctl_name = "Capture Mute",
.int_val = {off},
},
{
.ctl_name = "Left Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right PGA Mux",
.str_val = "DifferentialR",
},
{
.ctl_name = "Left PGA Mux",
.str_val = "DifferentialL",
},
{
.ctl_name = "Differential Mux",
.str_val = "Line 2",
},
};
const struct config_control es8388_hands_free_mic_capture_controls[] = {
{
.ctl_name = "Capture Digital Volume",
.int_val = {192, 192},
},
{
.ctl_name = "Capture Mute",
.int_val = {off},
},
{
.ctl_name = "Left Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right PGA Mux",
.str_val = "DifferentialR",
},
{
.ctl_name = "Left PGA Mux",
.str_val = "DifferentialL",
},
{
.ctl_name = "Differential Mux",
.str_val = "Line 1",
},
};
#ifndef _es8388_CONFIG_H_
#define _es8388_CONFIG_H_
#include "config.h"
const struct config_control es8388_speaker_normal_controls[] = {
{
.ctl_name = "Left Mixer Left Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Right Mixer Right Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Output 2 Playback Volume",
.int_val = {27, 27},
},
{
.ctl_name = "aw87xxx_profile_switch_0",
.str_val = "Music",
},
{
.ctl_name = "aw87xxx_profile_switch_1",
.str_val = "Music",
},
{
.ctl_name = "OUT2 Switch",
.int_val = {on},
},
};
const struct config_control es8388_speaker_incall_controls[] = {
};
const struct config_control es8388_speaker_ringtone_controls[] = {
};
const struct config_control es8388_speaker_voip_controls[] = {
};
const struct config_control es8388_earpiece_normal_controls[] = {
};
const struct config_control es8388_earpiece_incall_controls[] = {
};
const struct config_control es8388_earpiece_ringtone_controls[] = {
};
const struct config_control es8388_earpiece_voip_controls[] = {
};
const struct config_control es8388_headphone_normal_controls[] = {
{
.ctl_name = "Left Mixer Left Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Right Mixer Right Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Output 1 Playback Volume",
.int_val = {27, 27},
},
{
.ctl_name = "OUT1 Switch",
.int_val = {on},
},
};
const struct config_control es8388_headphone_incall_controls[] = {
};
const struct config_control es8388_headphone_ringtone_controls[] = {
};
const struct config_control es8388_speaker_headphone_normal_controls[] = {
{
.ctl_name = "Left Mixer Left Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Right Mixer Right Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Output 1 Playback Volume",
.int_val = {27, 27},
},
{
.ctl_name = "OUT1 Switch",
.int_val = {on},
},
};
const struct config_control es8388_speaker_headphone_ringtone_controls[] = {
};
const struct config_control es8388_headphone_voip_controls[] = {
};
const struct config_control es8388_headset_normal_controls[] = {
{
.ctl_name = "Left Mixer Left Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Right Mixer Right Playback Switch",
.int_val = {on},
},
{
.ctl_name = "Output 1 Playback Volume",
.int_val = {27, 27},
},
{
.ctl_name = "OUT1 Switch",
.int_val = {on},
},
};
const struct config_control es8388_headset_incall_controls[] = {
};
const struct config_control es8388_headset_ringtone_controls[] = {
};
const struct config_control es8388_headset_voip_controls[] = {
};
const struct config_control es8388_bluetooth_normal_controls[] = {
};
const struct config_control es8388_bluetooth_incall_controls[] = {
};
const struct config_control es8388_bluetooth_voip_controls[] = {
};
const struct config_control es8388_main_mic_capture_controls[] = {
{
.ctl_name = "Capture Digital Volume",
.int_val = {192, 192},
},
{
.ctl_name = "Capture Mute",
.int_val = {off},
},
{
.ctl_name = "Left Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right PGA Mux",
.str_val = "DifferentialR",
},
{
.ctl_name = "Left PGA Mux",
.str_val = "DifferentialL",
},
{
.ctl_name = "Differential Mux",
.str_val = "Line 2",
},
};
const struct config_control es8388_hands_free_mic_capture_controls[] = {
{
.ctl_name = "Capture Digital Volume",
.int_val = {192, 192},
},
{
.ctl_name = "Capture Mute",
.int_val = {off},
},
{
.ctl_name = "Left Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right Channel Capture Volume",
.int_val = {3},
},
{
.ctl_name = "Right PGA Mux",
.str_val = "DifferentialR",
},
{
.ctl_name = "Left PGA Mux",
.str_val = "DifferentialL",
},
{
.ctl_name = "Differential Mux",
.str_val = "Line 1",
},
};
const struct config_control es8388_bluetooth_sco_mic_capture_controls[] = {
};
const struct config_control es8388_playback_off_controls[] = {
{
.ctl_name = "OUT2 Switch",
.int_val = {off},
},
{
.ctl_name = "OUT1 Switch",
.int_val = {off},
},
{
.ctl_name = "Left Mixer Left Playback Switch",
.int_val = {off},
},
{
.ctl_name = "Right Mixer Right Playback Switch",
.int_val = {off},
},
{
.ctl_name = "Output 2 Playback Volume",
.int_val = {0, 0},
},
{
.ctl_name = "Output 1 Playback Volume",
.int_val = {0, 0},
},
{
.ctl_name = "aw87xxx_profile_switch_0",
.str_val = "Off",
},
{
.ctl_name = "aw87xxx_profile_switch_1",
.str_val = "Off",
},
};
const struct config_control es8388_capture_off_controls[] = {
{
.ctl_name = "Capture Mute",
.int_val = {on},
},
};
const struct config_control es8388_incall_off_controls[] = {
};
const struct config_control es8388_voip_off_controls[] = {
};
const struct config_route_table es8388_config_table = {
//speaker
.speaker_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_speaker_normal_controls,
.controls_count = sizeof(es8388_speaker_normal_controls) / sizeof(struct config_control),
},
.speaker_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_speaker_incall_controls,
.controls_count = sizeof(es8388_speaker_incall_controls) / sizeof(struct config_control),
},
.speaker_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_speaker_ringtone_controls,
.controls_count = sizeof(es8388_speaker_ringtone_controls) / sizeof(struct config_control),
},
.speaker_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_speaker_voip_controls,
.controls_count = sizeof(es8388_speaker_voip_controls) / sizeof(struct config_control),
},
//earpiece
.earpiece_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_earpiece_normal_controls,
.controls_count = sizeof(es8388_earpiece_normal_controls) / sizeof(struct config_control),
},
.earpiece_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_earpiece_incall_controls,
.controls_count = sizeof(es8388_earpiece_incall_controls) / sizeof(struct config_control),
},
.earpiece_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_earpiece_ringtone_controls,
.controls_count = sizeof(es8388_earpiece_ringtone_controls) / sizeof(struct config_control),
},
.earpiece_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_earpiece_voip_controls,
.controls_count = sizeof(es8388_earpiece_voip_controls) / sizeof(struct config_control),
},
//headphone
.headphone_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headphone_normal_controls,
.controls_count = sizeof(es8388_headphone_normal_controls) / sizeof(struct config_control),
},
.headphone_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headphone_incall_controls,
.controls_count = sizeof(es8388_headphone_incall_controls) / sizeof(struct config_control),
},
.headphone_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headphone_ringtone_controls,
.controls_count = sizeof(es8388_headphone_ringtone_controls) / sizeof(struct config_control),
},
.speaker_headphone_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_speaker_headphone_normal_controls,
.controls_count = sizeof(es8388_speaker_headphone_normal_controls) / sizeof(struct config_control),
},
.speaker_headphone_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_speaker_headphone_ringtone_controls,
.controls_count = sizeof(es8388_speaker_headphone_ringtone_controls) / sizeof(struct config_control),
},
.headphone_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headphone_voip_controls,
.controls_count = sizeof(es8388_headphone_voip_controls) / sizeof(struct config_control),
},
//headset
.headset_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headset_normal_controls,
.controls_count = sizeof(es8388_headset_normal_controls) / sizeof(struct config_control),
},
.headset_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headset_incall_controls,
.controls_count = sizeof(es8388_headset_incall_controls) / sizeof(struct config_control),
},
.headset_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headset_ringtone_controls,
.controls_count = sizeof(es8388_headset_ringtone_controls) / sizeof(struct config_control),
},
.headset_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_headset_voip_controls,
.controls_count = sizeof(es8388_headset_voip_controls) / sizeof(struct config_control),
},
//bluetooth
.bluetooth_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_bluetooth_normal_controls,
.controls_count = sizeof(es8388_bluetooth_normal_controls) / sizeof(struct config_control),
},
.bluetooth_incall = {
.sound_card = 0,
.devices = DEVICES_0_1,
.controls = es8388_bluetooth_incall_controls,
.controls_count = sizeof(es8388_bluetooth_incall_controls) / sizeof(struct config_control),
},
.bluetooth_voip = {
.sound_card = 0,
.devices = DEVICES_0_1,
.controls = es8388_bluetooth_voip_controls,
.controls_count = sizeof(es8388_bluetooth_voip_controls) / sizeof(struct config_control),
},
//capture
.main_mic_capture = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_main_mic_capture_controls,
.controls_count = sizeof(es8388_main_mic_capture_controls) / sizeof(struct config_control),
},
.hands_free_mic_capture = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = es8388_hands_free_mic_capture_controls,
.controls_count = sizeof(es8388_hands_free_mic_capture_controls) / sizeof(struct config_control),
},
.bluetooth_sco_mic_capture = {
.sound_card = 0,
.devices = DEVICES_0_1,
.controls = es8388_bluetooth_sco_mic_capture_controls,
.controls_count = sizeof(es8388_bluetooth_sco_mic_capture_controls) / sizeof(struct config_control),
},
//off
.playback_off = {
.controls = es8388_playback_off_controls,
.controls_count = sizeof(es8388_playback_off_controls) / sizeof(struct config_control),
},
.capture_off = {
.controls = es8388_capture_off_controls,
.controls_count = sizeof(es8388_capture_off_controls) / sizeof(struct config_control),
},
.incall_off = {
.controls = es8388_incall_off_controls,
.controls_count = sizeof(es8388_incall_off_controls) / sizeof(struct config_control),
},
.voip_off = {
.controls = es8388_voip_off_controls,
.controls_count = sizeof(es8388_voip_off_controls) / sizeof(struct config_control),
},
#ifdef BOX_HAL
//hdmi
.hdmi_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls_count = 0,
},
#else
//hdmi
.hdmi_normal = {
.sound_card = 1,
.devices = DEVICES_0,
.controls_count = 0,
},
#endif
//spdif
.spdif_normal = {
.sound_card = 1,
.devices = DEVICES_0,
.controls_count = 0,
},
//usb audio
.usb_normal = {
.sound_card = 2,
.devices = DEVICES_0,
.controls_count = 0,
},
.usb_capture = {
.sound_card = 2,
.devices = DEVICES_0,
.controls_count = 0,
},
};
#endif //_es8388_CONFIG_H_
3、【新建】
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\codec_config\nau8822_config.h
Y:\OK3576-android-source\hardware\rockchip\audio\tinyalsa_hal\codec_config\nau8822_config.h
#ifndef _nau8822_CONFIG_H_
#define _nau8822_CONFIG_H_
#include "config.h"
const struct config_control nau8822_speaker_normal_controls[] = {
{
.ctl_name = "Speaker Volume",
.int_val = {63,63},
},
{
.ctl_name = "Speaker Playback Switch",
.int_val = {on,on},
},
{
.ctl_name = "Speaker Switch",
.int_val = {on},
},
};
const struct config_control nau8822_speaker_incall_controls[] = {
};
const struct config_control nau8822_speaker_ringtone_controls[] = {
};
const struct config_control nau8822_speaker_voip_controls[] = {
};
const struct config_control nau8822_earpiece_normal_controls[] = {
};
const struct config_control nau8822_earpiece_incall_controls[] = {
};
const struct config_control nau8822_earpiece_ringtone_controls[] = {
};
const struct config_control nau8822_earpiece_voip_controls[] = {
};
const struct config_control nau8822_headphone_normal_controls[] = {
{
.ctl_name = "Headphone Playback Switch",
.int_val = {on, on},
},
{
.ctl_name = "Headphone Switch",
.int_val = {on},
},
{
.ctl_name = "Headphone Volume",
.int_val = {63,63},
},
};
const struct config_control nau8822_headphone_incall_controls[] = {
};
const struct config_control nau8822_headphone_ringtone_controls[] = {
};
const struct config_control nau8822_speaker_headphone_normal_controls[] = {
};
const struct config_control nau8822_speaker_headphone_ringtone_controls[] = {
};
const struct config_control nau8822_headphone_voip_controls[] = {
};
const struct config_control nau8822_headset_normal_controls[] = {
{
.ctl_name = "Headphone Playback Switch",
.int_val = {on, on},
},
{
.ctl_name = "Headphone Switch",
.int_val = {on},
},
};
const struct config_control nau8822_headset_incall_controls[] = {
};
const struct config_control nau8822_headset_ringtone_controls[] = {
};
const struct config_control nau8822_headset_voip_controls[] = {
};
const struct config_control nau8822_bluetooth_normal_controls[] = {
};
const struct config_control nau8822_bluetooth_incall_controls[] = {
};
const struct config_control nau8822_bluetooth_voip_controls[] = {
};
const struct config_control nau8822_main_mic_capture_controls[] = {
{
.ctl_name = "Main Mic Switch",
.int_val = {on},
},
{
.ctl_name = "Headset Mic Switch",
.int_val = {on},
},
};
const struct config_control nau8822_hands_free_mic_capture_controls[] = {
{
.ctl_name = "Main Mic Switch",
.int_val = {on},
},
{
.ctl_name = "Headset Mic Switch",
.int_val = {on},
},
};
const struct config_control nau8822_bluetooth_sco_mic_capture_controls[] = {
};
const struct config_control nau8822_playback_off_controls[] = {
{
.ctl_name = "Speaker Switch",
.int_val = {off},
},
{
.ctl_name = "Headphone Switch",
.int_val = {off},
},
};
const struct config_control nau8822_capture_off_controls[] = {
{
.ctl_name = "Main Mic Switch",
.int_val = {off},
},
{
.ctl_name = "Headset Mic Switch",
.int_val = {off},
},
};
const struct config_control nau8822_incall_off_controls[] = {
};
const struct config_control nau8822_voip_off_controls[] = {
};
const struct config_route_table nau8822_config_table = {
//speaker
.speaker_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_speaker_normal_controls,
.controls_count = sizeof(nau8822_speaker_normal_controls) / sizeof(struct config_control),
},
.speaker_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_speaker_incall_controls,
.controls_count = sizeof(nau8822_speaker_incall_controls) / sizeof(struct config_control),
},
.speaker_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_speaker_ringtone_controls,
.controls_count = sizeof(nau8822_speaker_ringtone_controls) / sizeof(struct config_control),
},
.speaker_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_speaker_voip_controls,
.controls_count = sizeof(nau8822_speaker_voip_controls) / sizeof(struct config_control),
},
//earpiece
.earpiece_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_earpiece_normal_controls,
.controls_count = sizeof(nau8822_earpiece_normal_controls) / sizeof(struct config_control),
},
.earpiece_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_earpiece_incall_controls,
.controls_count = sizeof(nau8822_earpiece_incall_controls) / sizeof(struct config_control),
},
.earpiece_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_earpiece_ringtone_controls,
.controls_count = sizeof(nau8822_earpiece_ringtone_controls) / sizeof(struct config_control),
},
.earpiece_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_earpiece_voip_controls,
.controls_count = sizeof(nau8822_earpiece_voip_controls) / sizeof(struct config_control),
},
//headphone
.headphone_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headphone_normal_controls,
.controls_count = sizeof(nau8822_headphone_normal_controls) / sizeof(struct config_control),
},
.headphone_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headphone_incall_controls,
.controls_count = sizeof(nau8822_headphone_incall_controls) / sizeof(struct config_control),
},
.headphone_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headphone_ringtone_controls,
.controls_count = sizeof(nau8822_headphone_ringtone_controls) / sizeof(struct config_control),
},
.speaker_headphone_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_speaker_headphone_normal_controls,
.controls_count = sizeof(nau8822_speaker_headphone_normal_controls) / sizeof(struct config_control),
},
.speaker_headphone_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_speaker_headphone_ringtone_controls,
.controls_count = sizeof(nau8822_speaker_headphone_ringtone_controls) / sizeof(struct config_control),
},
.headphone_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headphone_voip_controls,
.controls_count = sizeof(nau8822_headphone_voip_controls) / sizeof(struct config_control),
},
//headset
.headset_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headset_normal_controls,
.controls_count = sizeof(nau8822_headset_normal_controls) / sizeof(struct config_control),
},
.headset_incall = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headset_incall_controls,
.controls_count = sizeof(nau8822_headset_incall_controls) / sizeof(struct config_control),
},
.headset_ringtone = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headset_ringtone_controls,
.controls_count = sizeof(nau8822_headset_ringtone_controls) / sizeof(struct config_control),
},
.headset_voip = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_headset_voip_controls,
.controls_count = sizeof(nau8822_headset_voip_controls) / sizeof(struct config_control),
},
//bluetooth
.bluetooth_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_bluetooth_normal_controls,
.controls_count = sizeof(nau8822_bluetooth_normal_controls) / sizeof(struct config_control),
},
.bluetooth_incall = {
.sound_card = 0,
.devices = DEVICES_0_1,
.controls = nau8822_bluetooth_incall_controls,
.controls_count = sizeof(nau8822_bluetooth_incall_controls) / sizeof(struct config_control),
},
.bluetooth_voip = {
.sound_card = 0,
.devices = DEVICES_0_1,
.controls = nau8822_bluetooth_voip_controls,
.controls_count = sizeof(nau8822_bluetooth_voip_controls) / sizeof(struct config_control),
},
//capture
.main_mic_capture = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_main_mic_capture_controls,
.controls_count = sizeof(nau8822_main_mic_capture_controls) / sizeof(struct config_control),
},
.hands_free_mic_capture = {
.sound_card = 0,
.devices = DEVICES_0,
.controls = nau8822_hands_free_mic_capture_controls,
.controls_count = sizeof(nau8822_hands_free_mic_capture_controls) / sizeof(struct config_control),
},
.bluetooth_sco_mic_capture = {
.sound_card = 0,
.devices = DEVICES_0_1,
.controls = nau8822_bluetooth_sco_mic_capture_controls,
.controls_count = sizeof(nau8822_bluetooth_sco_mic_capture_controls) / sizeof(struct config_control),
},
//off
.playback_off = {
.controls = nau8822_playback_off_controls,
.controls_count = sizeof(nau8822_playback_off_controls) / sizeof(struct config_control),
},
.capture_off = {
.controls = nau8822_capture_off_controls,
.controls_count = sizeof(nau8822_capture_off_controls) / sizeof(struct config_control),
},
.incall_off = {
.controls = nau8822_incall_off_controls,
.controls_count = sizeof(nau8822_incall_off_controls) / sizeof(struct config_control),
},
.voip_off = {
.controls = nau8822_voip_off_controls,
.controls_count = sizeof(nau8822_voip_off_controls) / sizeof(struct config_control),
},
#ifdef BOX_HAL
//hdmi
.hdmi_normal = {
.sound_card = 0,
.devices = DEVICES_0,
.controls_count = 0,
},
#else
//hdmi
.hdmi_normal = {
.sound_card = 1,
.devices = DEVICES_0,
.controls_count = 0,
},
#endif
//spdif
.spdif_normal = {
.sound_card = 1,
.devices = DEVICES_0,
.controls_count = 0,
},
//usb audio
.usb_normal = {
.sound_card = 2,
.devices = DEVICES_0,
.controls_count = 0,
},
.usb_capture = {
.sound_card = 2,
.devices = DEVICES_0,
.controls_count = 0,
},
};
#endif //_nau8822_CONFIG_H_
4、
Y:\OK3576-android-source\hardware\rockchip\audio\tinyalsa_hal\audio_hw.c
Y:\rk3576-android14.0-20250217\hardware\rockchip\audio\tinyalsa_hal\audio_hw.c
/*
* Copyright (C) 2012 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.
*/
/**
* @file audio_hw.c
* @brief
* ALSA Audio Git Log
* - V0.1.0:add alsa audio hal,just support 312x now.
* - V0.2.0:remove unused variable.
* - V0.3.0:turn off device when do_standby.
* - V0.4.0:turn off device before open pcm.
* - V0.4.1:Need to re-open the control to fix no sound when suspend.
* - V0.5.0:Merge the mixer operation from legacy_alsa.
* - V0.6.0:Merge speex denoise from legacy_alsa.
* - V0.7.0:add copyright.
* - V0.7.1:add support for box audio
* - V0.7.2:add support for dircet output
* - V0.8.0:update the direct output for box, add the DVI mode
* - V1.0.0:stable version
*
* @author RkAudio
* @version 1.0.5
* @date 2015-08-24
*/
//#define LOG_NDEBUG 0
#ifdef PRIMARY_HAL
#define LOG_TAG "modules.primary.audio_hal"
#endif
#ifdef EXT_1_HAL
#define LOG_TAG "modules.ext_1.audio_hal"
#endif
#ifdef EXT_2_HAL
#define LOG_TAG "modules.ext_2.audio_hal"
#endif
#ifdef EXT_3_HAL
#define LOG_TAG "modules.ext_3.audio_hal"
#endif
#ifdef EXT_4_HAL
#define LOG_TAG "modules.ext_4.audio_hal"
#endif
#include <system/audio.h>
#include "audio_hw.h"
#include "alsa_mixer.h"
#include "alsa_route.h"
#include "voice/voice.h"
#include "codec_config/config.h"
#include "utils/audio_time.h"
#include "utils/channel.h"
#include <inttypes.h>
#include <unistd.h>
#include <fcntl.h>
#include <ctype.h>
#define SNDRV_CARDS 8
#define SNDRV_DEVICES 8
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
#define SND_CARDS_NODE "/proc/asound/cards"
#define SAMPLECOUNT 441*5*2*2
#define HDMI_BITSTREAM_BYPASS "ELD Bypass Switch"
#define PCM_CAPTURE_CHANNELS 2
// #define PCM_REFERENCE_CHANNELS 2
static struct pcm_config pcm_config = {
.channels = 2,
.rate = 48000,
.period_size = 480,
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
static struct pcm_config pcm_config_in = {
#if PCM_REFERENCE_CHANNELS
.channels = PCM_CAPTURE_CHANNELS + PCM_REFERENCE_CHANNELS,
#else
.channels = PCM_CAPTURE_CHANNELS,
#endif
.rate = 48000,
.period_size = 480, // 10ms
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
static struct pcm_config pcm_config_in_low_latency = {
.channels = 2,
.rate = 48000,
.period_size = 256,
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
static struct pcm_config pcm_config_sco = {
.channels = 1,
.rate = 8000,
.period_size = 128,
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
/* for bt client call*/
static struct pcm_config pcm_config_hfp = {
.channels = 2,
.rate = 48000,
.period_size = 256,
.period_count = 4,
};
static struct pcm_config pcm_config_ap_sco = {
.channels = 2,
.rate = 8000,
.period_size = 80,
.period_count = 4,
};
static struct pcm_config pcm_config_in_bt = {
.channels = 2,
.rate = 8000,
.period_size = 120,
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
static struct pcm_config pcm_config_deep = {
.channels = 2,
.rate = 48000,
/* FIXME This is an arbitrary number, may change.
* Dynamic configuration based on screen on/off is not implemented;
* let's see what power consumption is first to see if necessary.
*/
.period_size = 8192,
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
static struct pcm_config pcm_config_hdmi_multi = {
.channels = 6, /* changed when the stream is opened */
.rate = HDMI_MULTI_DEFAULT_SAMPLING_RATE,
.period_size = 1024,
.period_count = 4,
.format = PCM_FORMAT_S16_LE,
};
static struct pcm_config pcm_config_direct = {
.channels = 2,
.rate = 48000,
.period_size = 1024,
.period_count = 4,
#ifdef IEC958_FORAMT
.format = PCM_FORMAT_IEC958_SUBFRAME_LE,
#else
.format = PCM_FORMAT_S24_LE,
#endif
};
static const struct route_config media_speaker = {
"media-speaker",
"media-main-mic",
"playback-off",
"capture-off",
};
static const struct route_config media_headphones = {
"media-headphones",
"media-main-mic",
"playback-off",
"capture-off",
};
static const struct route_config media_headset = {
"media-headphones",
"media-headset-mic",
"playback-off",
"capture-off",
};
static const struct route_config camcorder_speaker = {
"media-speaker",
"media-second-mic",
"playback-off",
"capture-off",
};
static const struct route_config camcorder_headphones = {
"media-headphones",
"media-second-mic",
"playback-off",
"capture-off",
};
static const struct route_config voice_rec_speaker = {
"voice-rec-speaker",
"voice-rec-main-mic",
"incall-off",
"incall-off",
};
static const struct route_config voice_rec_headphones = {
"voice-rec-headphones",
"voice-rec-main-mic",
"incall-off",
"incall-off",
};
static const struct route_config voice_rec_headset = {
"voice-rec-headphones",
"voice-rec-headset-mic",
"incall-off",
"incall-off",
};
static const struct route_config communication_speaker = {
"communication-speaker",
"communication-main-mic",
"voip-off",
"voip-off",
};
static const struct route_config communication_headphones = {
"communication-headphones",
"communication-main-mic",
"voip-off",
"voip-off",
};
static const struct route_config communication_headset = {
"communication-headphones",
"communication-headset-mic",
"voip-off",
"voip-off",
};
static const struct route_config speaker_and_headphones = {
"speaker-and-headphones",
"main-mic",
"playback-off",
"capture-off",
};
static const struct route_config bluetooth_sco = {
"bt-sco-headset",
"bt-sco-mic",
"playback-off",
"capture-off",
};
static const struct route_config * const route_configs[IN_SOURCE_TAB_SIZE]
[OUT_DEVICE_TAB_SIZE] = {
{ /* IN_SOURCE_MIC */
&media_speaker, /* OUT_DEVICE_SPEAKER */
&media_headset, /* OUT_DEVICE_HEADSET */
&media_headphones, /* OUT_DEVICE_HEADPHONES */
&bluetooth_sco, /* OUT_DEVICE_BT_SCO */
&speaker_and_headphones /* OUT_DEVICE_SPEAKER_AND_HEADSET */
},
{ /* IN_SOURCE_CAMCORDER */
&camcorder_speaker, /* OUT_DEVICE_SPEAKER */
&camcorder_headphones, /* OUT_DEVICE_HEADSET */
&camcorder_headphones, /* OUT_DEVICE_HEADPHONES */
&bluetooth_sco, /* OUT_DEVICE_BT_SCO */
&speaker_and_headphones /* OUT_DEVICE_SPEAKER_AND_HEADSET */
},
{ /* IN_SOURCE_VOICE_RECOGNITION */
&voice_rec_speaker, /* OUT_DEVICE_SPEAKER */
&voice_rec_headset, /* OUT_DEVICE_HEADSET */
&voice_rec_headphones, /* OUT_DEVICE_HEADPHONES */
&bluetooth_sco, /* OUT_DEVICE_BT_SCO */
&speaker_and_headphones /* OUT_DEVICE_SPEAKER_AND_HEADSET */
},
{ /* IN_SOURCE_VOICE_COMMUNICATION */
&communication_speaker, /* OUT_DEVICE_SPEAKER */
&communication_headset, /* OUT_DEVICE_HEADSET */
&communication_headphones, /* OUT_DEVICE_HEADPHONES */
&bluetooth_sco, /* OUT_DEVICE_BT_SCO */
&speaker_and_headphones /* OUT_DEVICE_SPEAKER_AND_HEADSET */
}
};
struct SurroundFormat {
audio_format_t format;
const char *value;
};
const struct SurroundFormat sSurroundFormat[] = {
{AUDIO_FORMAT_AC3,"AUDIO_FORMAT_AC3"},
{AUDIO_FORMAT_E_AC3,"AUDIO_FORMAT_E_AC3"},
{AUDIO_FORMAT_DTS,"AUDIO_FORMAT_DTS"},
{AUDIO_FORMAT_DTS_HD,"AUDIO_FORMAT_DTS_HD"},
{AUDIO_FORMAT_AAC_LC,"AUDIO_FORMAT_AAC_LC"},
{AUDIO_FORMAT_DOLBY_TRUEHD,"AUDIO_FORMAT_DOLBY_TRUEHD"},
{AUDIO_FORMAT_AC4,"AUDIO_FORMAT_E_AC3_JOC"}
};
enum SOUND_CARD_OWNER{
SOUND_CARD_HDMI = 0,
SOUND_CARD_SPDIF = 1,
};
/*
* mute audio datas when screen off or standby
* The MediaPlayer no stop/pause when screen off, they may be just play in background,
* so they still send audio datas to audio hal.
* HDMI may disconnet and enter stanby status, this means no voice output on HDMI
* but speaker/av and spdif still work, and voice may output on them.
* Some customer need to mute the audio datas in this condition.
* If need mute datas when screen off, define this marco.
*/
//#define MUTE_WHEN_SCREEN_OFF
/*
* if current audio stream bitstream over hdmi,
* and hdmi is removed and reconnected later,
* the driver of hdmi may config it with pcm mode automatically,
* which is according the implement of hdmi driver.
* If hdmi driver implement in this way, in order to output audio
* bitstream stream after hdmi reconnected,
* we must close sound card of hdmi and reopen/config
* it in bitstream mode. If need this, define this macro.
*/
#define AUDIO_BITSTREAM_REOPEN_HDMI
//#define ALSA_DEBUG
#ifdef ALSA_IN_DEBUG
FILE *in_debug;
#endif
int in_dump(const struct audio_stream *stream, int fd);
int out_dump(const struct audio_stream *stream, int fd);
/**
* @brief get_output_device_id
*
* @param device
*
* @returns
*/
int get_output_device_id(audio_devices_t device)
{
if (device == AUDIO_DEVICE_NONE)
return OUT_DEVICE_NONE;
if (popcount(device) == 2) {
if ((device == (AUDIO_DEVICE_OUT_SPEAKER |
AUDIO_DEVICE_OUT_WIRED_HEADSET)) ||
(device == (AUDIO_DEVICE_OUT_SPEAKER |
AUDIO_DEVICE_OUT_WIRED_HEADPHONE)))
return OUT_DEVICE_SPEAKER_AND_HEADSET;
else
return OUT_DEVICE_NONE;
}
if (popcount(device) != 1)
return OUT_DEVICE_NONE;
switch (device) {
case AUDIO_DEVICE_OUT_SPEAKER:
return OUT_DEVICE_SPEAKER;
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
return OUT_DEVICE_HEADSET;
case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
return OUT_DEVICE_HEADPHONES;
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
return OUT_DEVICE_BT_SCO;
default:
return OUT_DEVICE_NONE;
}
}
/**
* @brief get_input_source_id
*
* @param source
*
* @returns
*/
int get_input_source_id(audio_source_t source)
{
switch (source) {
case AUDIO_SOURCE_DEFAULT:
return IN_SOURCE_NONE;
case AUDIO_SOURCE_MIC:
return IN_SOURCE_MIC;
case AUDIO_SOURCE_CAMCORDER:
return IN_SOURCE_CAMCORDER;
case AUDIO_SOURCE_VOICE_RECOGNITION:
return IN_SOURCE_VOICE_RECOGNITION;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
return IN_SOURCE_VOICE_COMMUNICATION;
default:
return IN_SOURCE_NONE;
}
}
/**
* @brief force_non_hdmi_out_standby
* must be called with hw device outputs list, all out streams, and hw device mutexes locked
*
* @param adev
*/
static void force_non_hdmi_out_standby(struct audio_device *adev)
{
enum output_type type;
struct stream_out *out;
for (type = 0; type < OUTPUT_TOTAL; ++type) {
out = adev->outputs[type];
if (type == OUTPUT_HDMI_MULTI|| !out)
continue;
/* This will never recurse more than 2 levels deep. */
do_out_standby(out);
}
}
/**
* @brief getOutputRouteFromDevice
*
* @param device
*
* @returns
*/
unsigned getOutputRouteFromDevice(uint32_t device)
{
switch (device) {
case AUDIO_DEVICE_OUT_SPEAKER:
return SPEAKER_NORMAL_ROUTE;
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
return HEADSET_NORMAL_ROUTE;
case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
return HEADPHONE_NORMAL_ROUTE;
case (AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADPHONE):
case (AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET):
return SPEAKER_HEADPHONE_NORMAL_ROUTE;
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
return BLUETOOTH_NORMAL_ROUTE;
case AUDIO_DEVICE_OUT_AUX_DIGITAL:
#ifdef SUPPORT_VX_ROCKCHIP
case VX_ROCKCHIP_OUT_HDMI0:
#endif
return HDMI_NORMAL_ROUTE;
default:
return PLAYBACK_OFF_ROUTE;
}
}
/**
* @brief getVoiceRouteFromDevice
*
* @param device
*
* @returns
*/
uint32_t getVoiceRouteFromDevice(uint32_t device)
{
ALOGE("not support now");
return 0;
}
/**
* @brief getInputRouteFromDevice
*
* @param device
*
* @returns
*/
uint32_t getInputRouteFromDevice(uint32_t device)
{
/*if (mMicMute) {
return CAPTURE_OFF_ROUTE;
}*/
ALOGE("%s:device:%x",__FUNCTION__,device);
switch (device) {
case AUDIO_DEVICE_IN_BUILTIN_MIC:
return MAIN_MIC_CAPTURE_ROUTE;
case AUDIO_DEVICE_IN_WIRED_HEADSET:
return HANDS_FREE_MIC_CAPTURE_ROUTE;
case AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET:
return BLUETOOTH_SOC_MIC_CAPTURE_ROUTE;
case AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET:
return USB_CAPTURE_ROUTE;
case AUDIO_DEVICE_IN_HDMI:
return HDMI_IN_CAPTURE_ROUTE;
default:
return CAPTURE_OFF_ROUTE;
}
}
/**
* @brief getRouteFromDevice
*
* @param device
*
* @returns
*/
uint32_t getRouteFromDevice(uint32_t device)
{
if (device & AUDIO_DEVICE_BIT_IN)
return getInputRouteFromDevice(device);
else
return getOutputRouteFromDevice(device);
}
static struct stream_out* adev_get_stream_out_by_io_handle_l(
struct audio_device* adev, audio_io_handle_t handle) {
struct listnode *node;
list_for_each (node, &adev->output_stream_list) {
struct stream_out *out = node_to_item(node, struct stream_out, list_node);
if (out && out->handle == handle) {
return out;
}
}
return NULL;
}
bool is_type_in_outdevices(struct stream_out *out, audio_devices_t type) {
for (int i = 0; i < out->num_configs; ++i)
{
if (out->devices[i] == type) {
return true;
}
}
return false;
}
static struct stream_in* adev_get_stream_in_by_io_handle_l(
struct audio_device* adev, audio_io_handle_t handle) {
struct listnode *node;
list_for_each (node, &adev->input_stream_list) {
struct stream_in *in = node_to_item(node, struct stream_in, list_node);
if (in && in->handle == handle) {
return in;
}
}
return NULL;
}
static struct stream_out* adev_get_stream_out_by_patch_handle_l(
struct audio_device* adev, audio_patch_handle_t patch_handle) {
struct listnode *node;
list_for_each (node, &adev->output_stream_list) {
struct stream_out *out = node_to_item(node, struct stream_out, list_node);
if (out && out->patch_handle == patch_handle) {
return out;
}
}
return NULL;
}
static void device_lock(struct audio_device *adev) {
pthread_mutex_lock(&adev->lock);
}
static int device_try_lock(struct audio_device *adev) {
return pthread_mutex_trylock(&adev->lock);
}
static void device_unlock(struct audio_device *adev) {
pthread_mutex_unlock(&adev->lock);
}
static struct stream_in* adev_get_stream_in_by_patch_handle_l(
struct audio_device* adev, audio_patch_handle_t patch_handle) {
struct listnode *node;
list_for_each (node, &adev->input_stream_list) {
struct stream_in *in = node_to_item(node, struct stream_in, list_node);
if (in && in->patch_handle == patch_handle) {
return in;
}
}
return NULL;
}
/*
* streams list management
*/
static void adev_add_stream_to_list(
struct audio_device* adev, struct listnode* list, struct listnode* stream_node) {
device_lock(adev);
list_add_tail(list, stream_node);
device_unlock(adev);
}
struct dev_proc_info SPEAKER_OUT_NAME[] = /* add codes& dai name here*/
{
{"rockchipcarrk33", NULL},
{"realtekrt5616c", NULL,},
{"realtekrt5651co", "rt5651-aif1",},
{"realtekrt5670c", NULL,},
{"realtekrt5672c", NULL,},
{"realtekrt5678co", NULL,},
{"rkhdmianalogsnd", NULL,},
{"rockchipcx2072x", NULL,},
{"rockchipes8316c", NULL,},
{"rockchipes8323c", NULL,},
{"rockchipes8388c", NULL,},
{"rockchipes8388", NULL,},
{"rockchipes8396c", NULL,},
{"rockchipnau8822", NULL,},
{"rockchiprk", NULL, },
{"rockchiprk809co", NULL,},
{"rockchiprk817co", NULL,},
{"rockchiprt5640c", "rt5640-aif1",},
{"rockchiprt5670c", NULL,},
{"rockchiprt5672c", NULL,},
{"rockchipaw882xx", NULL,},
{"rk3528acodec", NULL},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info HDMI_OUT_NAME[] =
{
{"realtekrt5651co", "i2s-hifi",},
{"realtekrt5670co", "i2s-hifi",},
{"rkhdmidpsound", NULL,},
{"hdmisound", NULL},
{"rockchiphdmi", NULL,},
{"rockchiphdmi0", NULL,},
{"rockchiprt5640c", "i2s-hifi",},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info HDMI_1_OUT_NAME[] =
{
{"rockchiphdmi1", NULL,},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info HDMI_ARC_OUT_NAME[] =
{
{"rockchiphdmiarc", NULL,},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info SPDIF_OUT_NAME[] =
{
{"ROCKCHIPSPDIF", "dit-hifi",},
{"rockchipspdif", NULL,},
{"rockchipcdndp", NULL,},
{"rockchipdp0", NULL,},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info SPDIF_1_OUT_NAME[] =
{
{"rockchipdp1", NULL,},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info BT_OUT_NAME[] =
{
{"rockchipbt", NULL,},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info MIC_IN_NAME[] =
{
{"rockchipcarrk33", NULL},
{"realtekrt5616c", NULL,},
{"realtekrt5651co", "rt5651-aif1",},
{"realtekrt5670c", NULL,},
{"realtekrt5672c", NULL,},
{"realtekrt5678co", NULL,},
{"rockchipes8316c", NULL,},
{"rockchipes8323c", NULL,},
{"rockchipes8388c", NULL,},
{"rockchipes8388", NULL,},
{"rockchipes8396c", NULL,},
{"rockchipes7210", NULL,},
{"rockchipes7243", NULL,},
{"rockchipnau8822", NULL,},
{"rockchiprk", NULL, },
{"rockchiprk809co", NULL,},
{"rockchiprk817co", NULL,},
{"rockchiprt5640c", NULL,},
{"rockchiprt5670c", NULL,},
{"rockchiprt5672c", NULL,},
{"rockchippdmmica", NULL},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info HDMI_IN_NAME[] =
{
{"realtekrt5651co", "tc358749x-audio"},
{"hdmiin", NULL},
{"rockchiphdmirx", NULL},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info HDMI_IN1_NAME[] =
{
{"rockchiprk628hd", NULL},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
struct dev_proc_info BT_IN_NAME[] =
{
{"rockchipbt", NULL},
{NULL, NULL}, /* Note! Must end with NULL, else will cause crash */
};
static int name_match(const char* dst, const char* src)
{
int score = 0;
// total equal
if (!strcmp(dst, src)) {
score = 100;
} else if (strstr(dst, src)) {
// part equal
score = 50;
}
return score;
}
static bool is_specified_out_sound_card(char *id, struct dev_proc_info *match)
{
int i = 0;
if (!match)
return true; /* match any */
while (match[i].cid) {
if (!strcmp(id, match[i].cid)) {
return true;
}
i++;
}
return false;
}
static bool dev_id_match(const char *info, const char *did)
{
const char *deli = "id:";
char *id;
int idx = 0;
if (!did)
return true;
if (!info)
return false;
/* find str like-> id: ff880000.i2s-rt5651-aif1 rt5651-aif1-0 */
id = strstr(info, deli);
if (!id)
return false;
id += strlen(deli);
while(id[idx] != '\0') {
if (id[idx] == '\r' ||id[idx] == '\n') {
id[idx] = '\0';
break;
}
idx ++;
}
if (strstr(id, did)) {
ALOGE("match dai!!!: %s %s", id, did);
return true;
}
return false;
}
static bool get_specified_out_dev(struct dev_info *devinfo,
int card,
const char *id,
struct dev_proc_info *match)
{
int i = 0;
int device;
char str_device[32];
char info[256];
size_t len;
FILE* file = NULL;
int score = 0;
int better = devinfo->score;
int index = -1;
/* parse card id */
if (!match)
return true; /* match any */
while (match[i].cid) {
score = name_match(id, match[i].cid);
if (score > better) {
better = score;
index = i;
}
i++;
}
if (index < 0)
return false;
if (!match[index].cid)
return false;
if (!match[index].did) { /* no exist dai info, exit */
devinfo->card = card;
devinfo->device = 0;
devinfo->score = better;
ALOGD("%s card, got card=%d,device=%d", devinfo->id,
devinfo->card, devinfo->device);
return true;
}
/* parse device id */
for (device = 0; device < SNDRV_DEVICES; device++) {
sprintf(str_device, "proc/asound/card%d/pcm%dp/info", card, device);
if (access(str_device, 0)) {
continue;
}
file = fopen(str_device, "r");
if (!file) {
ALOGD("Could reading %s property", str_device);
continue;
}
len = fread(info, sizeof(char), sizeof(info)/sizeof(char), file);
fclose(file);
if (len == 0 || len > sizeof(info)/sizeof(char))
continue;
if (info[len - 1] == '\n') {
len--;
info[len] = '\0';
}
/* parse device dai */
if (dev_id_match(info, match[index].did)) {
devinfo->card = card;
devinfo->device = device;
devinfo->score = better;
ALOGD("%s card, got card=%d,device=%d", devinfo->id,
devinfo->card, devinfo->device);
return true;
}
}
return false;
}
static bool get_specified_in_dev(struct dev_info *devinfo,
int card,
const char *id,
struct dev_proc_info *match)
{
int i = 0;
int device;
char str_device[32];
char info[256];
size_t len;
FILE* file = NULL;
int score = 0;
int better = devinfo->score;
int index = -1;
/* parse card id */
if (!match)
return true; /* match any */
while (match[i].cid) {
score = name_match(id, match[i].cid);
if (score > better) {
better = score;
index = i;
}
i++;
}
if (index < 0)
return false;
if (!match[index].cid)
return false;
if (!match[index].did) { /* no exist dai info, exit */
devinfo->card = card;
devinfo->device = 0;
devinfo->score = better;
ALOGD("%s card, got card=%d,device=%d", devinfo->id,
devinfo->card, devinfo->device);
return true;
}
/* parse device id */
for (device = 0; device < SNDRV_DEVICES; device++) {
sprintf(str_device, "proc/asound/card%d/pcm%dc/info", card, device);
if (access(str_device, 0)) {
continue;
}
file = fopen(str_device, "r");
if (!file) {
ALOGD("Could reading %s property", str_device);
continue;
}
len = fread(info, sizeof(char), sizeof(info)/sizeof(char), file);
fclose(file);
if (len == 0 || len > sizeof(info)/sizeof(char))
continue;
if (info[len - 1] == '\n') {
len--;
info[len] = '\0';
}
/* parse device dai */
if (dev_id_match(info, match[i].did)) {
devinfo->card = card;
devinfo->device = device;
devinfo->score = better;
ALOGD("%s card, got card=%d,device=%d", devinfo->id,
devinfo->card, devinfo->device);
return true;
}
}
return false;
}
static bool is_specified_in_sound_card(char *id, struct dev_proc_info *match)
{
int i = 0;
/*
* mic: diffrent product may have diffrent card name,modify codes here
* for example: 0 [rockchiprk3328 ]: rockchip-rk3328 - rockchip-rk3328
*/
if (!match)
return true;/* match any */
while (match[i].cid) {
if (!strcmp(id, match[i].cid)) {
return true;
}
i++;
}
return false;
}
static void set_default_dev_info( struct dev_info *info, int size, int rid)
{
for(int i =0; i < size; i++) {
if (rid) {
info[i].id = NULL;
}
info[i].card = (int)SND_OUT_SOUND_CARD_UNKNOWN;
info[i].score = 0;
}
}
static void dumpdev_info(const char *tag, struct dev_info *devinfo, int count)
{
ALOGD("dump %s device info", tag);
for(int i = 0; i < count; i++) {
if (devinfo[i].id && devinfo[i].card != SND_OUT_SOUND_CARD_UNKNOWN)
ALOGD("dev_info %s card=%d, device:%d", devinfo[i].id,
devinfo[i].card,
devinfo[i].device);
}
}
/*
* get sound card infor by parser node: /proc/asound/cards
* the sound card number is not always the same value
*/
static void read_out_sound_card(struct stream_out *out)
{
struct audio_device *device = NULL;
int card = 0;
char str[32];
char id[20];
size_t len;
FILE* file = NULL;
if((out == NULL) || (out->dev == NULL)) {
return ;
}
device = out->dev;
set_default_dev_info(device->dev_out, SND_OUT_SOUND_CARD_UNKNOWN, 0);
for (card = 0; card < SNDRV_CARDS; card++) {
sprintf(str, "proc/asound/card%d/id", card);
if (access(str, 0)) {
continue;
}
file = fopen(str, "r");
if (!file) {
ALOGD("Could reading %s property", str);
continue;
}
len = fread(id, sizeof(char), sizeof(id)/sizeof(char), file);
fclose(file);
if (len == 0 || len > sizeof(id)/sizeof(char))
continue;
if (id[len - 1] == '\n') {
len--;
id[len] = '\0';
}
ALOGD("card%d id:%s", card, id);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_SPEAKER], card, id, SPEAKER_OUT_NAME);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_HDMI], card, id, HDMI_OUT_NAME);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_HDMI_1], card, id, HDMI_1_OUT_NAME);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_HDMI_ARC], card, id, HDMI_ARC_OUT_NAME);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_SPDIF], card, id, SPDIF_OUT_NAME);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_SPDIF_1], card, id, SPDIF_1_OUT_NAME);
get_specified_out_dev(&device->dev_out[SND_OUT_SOUND_CARD_BT], card, id, BT_OUT_NAME);
}
dumpdev_info("out", device->dev_out, SND_OUT_SOUND_CARD_MAX);
return ;
}
#ifdef PRIMARY_HAL
/*
* check whether hdmi/dp sound card registered or not
*/
static bool check_out_card_registered(struct dev_proc_info *info)
{
bool ret = false;
int card = 0;
char str[32];
char id[20];
size_t len;
FILE* file = NULL;
struct dev_info devInfo = {
.id = NULL,
.card = (int)SND_OUT_SOUND_CARD_UNKNOWN,
.device = 0,
.score = 90,
};
for (card = 0; card < SNDRV_CARDS; card++) {
sprintf(str, "proc/asound/card%d/id", card);
if (access(str, 0)) {
continue;
}
file = fopen(str, "r");
if (!file) {
ALOGD("Could reading %s property", str);
continue;
}
len = fread(id, sizeof(char), sizeof(id)/sizeof(char), file);
fclose(file);
if (len == 0 || len > sizeof(id)/sizeof(char))
continue;
if (id[len - 1] == '\n') {
len--;
id[len] = '\0';
}
ALOGD("card%d id:%s", card, id);
ret = get_specified_out_dev(&devInfo, card, id, info);
if (ret)
break;
}
return ret;
}
#endif
/*
* get sound card infor by parser node: /proc/asound/cards
* the sound card number is not always the same value
*/
static void read_in_sound_card(struct stream_in *in)
{
struct audio_device *device = NULL;
int card = 0;
char str[32];
char id[20];
size_t len;
FILE* file = NULL;
if((in == NULL) || (in->dev == NULL)){
return ;
}
device = in->dev;
set_default_dev_info(device->dev_in, SND_IN_SOUND_CARD_UNKNOWN, 0);
for (card = 0; card < SNDRV_CARDS; card++) {
sprintf(str, "proc/asound/card%d/id", card);
if(access(str, 0)) {
continue;
}
file = fopen(str, "r");
if (!file) {
ALOGD("Could reading %s property", str);
continue;
}
len = fread(id, sizeof(char), sizeof(id)/sizeof(char), file);
fclose(file);
if (len == 0 || len > sizeof(id)/sizeof(char))
continue;
if (id[len - 1] == '\n') {
len--;
id[len] = '\0';
}
get_specified_in_dev(&device->dev_in[SND_IN_SOUND_CARD_MIC], card, id, MIC_IN_NAME);
/* set HDMI audio input info if need hdmi audio input */
get_specified_in_dev(&device->dev_in[SND_IN_SOUND_CARD_HDMI], card, id, HDMI_IN_NAME);
get_specified_in_dev(&device->dev_in[SND_IN_SOUND_CARD_HDMI_1], card, id, HDMI_IN1_NAME);
get_specified_in_dev(&device->dev_in[SND_IN_SOUND_CARD_BT], card, id, BT_IN_NAME);
}
dumpdev_info("in", device->dev_in, SND_IN_SOUND_CARD_MAX);
return ;
}
static bool is_bitstream(struct stream_out *out)
{
if (out == NULL) {
return false;
}
if (out->config.format == PCM_FORMAT_IEC958_SUBFRAME_LE)
return true;
bool bitstream = false;
if (out->output_direct) {
switch(out->output_direct_mode) {
case HBR:
case NLPCM:
bitstream = true;
break;
case LPCM:
default:
bitstream = false;
break;
}
} else {
if(out->output_direct_mode != LPCM) {
ALOGD("%s: %d: error output_direct = false, but output_direct_mode != LPCM, this is error config",__FUNCTION__,__LINE__);
}
}
return bitstream;
}
static bool is_multi_pcm(struct stream_out *out)
{
if (out == NULL) {
return false;
}
bool multi = false;
if (out->output_direct && (out->output_direct_mode == LPCM) && (out->config.channels > 2)) {
multi = true;
}
return multi;
}
/**
* @brief mixer_hdmi_set_force_bypass
* force hdmi to bypass even if hdmi not support bypass
*/
static int mixer_hdmi_set_force_bypass(struct stream_out *out, int card)
{
int ret = 0;
struct mixer *pMixer = NULL;
struct mixer_ctl *pctl;
struct audio_device *adev = out->dev;
if (is_type_in_outdevices(out, AUDIO_DEVICE_OUT_AUX_DIGITAL) && card >= 0) {
pMixer = mixer_open_legacy(card);
if (!pMixer) {
return ret;
}
pctl = mixer_get_control(pMixer, HDMI_BITSTREAM_BYPASS, 0);
if (pctl != NULL) {
// do not care edid
if (is_bitstream(out)) {
ret = mixer_ctl_set_val(pctl, 1);
} else {
ret = mixer_ctl_set_val(pctl, 0);
}
}
mixer_close_legacy(pMixer);
}
return ret;
}
/**
* @brief mixer_mode_set
* for rk3399 audio output mixer mode set
* @param out
*
* @return
*/
static int mixer_mode_set(struct stream_out *out)
{
int ret = 0;
struct mixer *pMixer = NULL;
struct mixer_ctl *pctl;
struct audio_device *adev = out->dev;
/*
* set audio mode for hdmi
* The driver of hdmi read the audio mode to identify
* the type of audio stream according to audio mode.
* 1) LPCM: the stream is pcm format
* 2) NLPCM: the stream is bitstream format, AC3/EAC3/DTS use this format
* 3) HDR: the stream is bitstream format, TrueHD/Atoms/DTS-HD/DTS-X use this format.
*/
if (is_type_in_outdevices(out, AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
pMixer = mixer_open_legacy(adev->dev_out[SND_OUT_SOUND_CARD_HDMI].card);
if (!pMixer) {
ALOGE("mMixer is a null point %s %d,CARD = %d",__func__, __LINE__,adev->dev_out[SND_OUT_SOUND_CARD_HDMI].card);
return ret;
}
pctl = mixer_get_control(pMixer,"AUDIO MODE",0 );
if (pctl != NULL) {
ALOGD("Now set mixer audio_mode is %d for drm",out->output_direct_mode);
switch (out->output_direct_mode) {
case HBR:
ret = mixer_ctl_set_val(pctl , out->output_direct_mode);
break;
case NLPCM:
ret = mixer_ctl_set_val(pctl , out->output_direct_mode);
break;
default:
ret = mixer_ctl_set_val(pctl , out->output_direct_mode);
break;
}
if (ret != 0) {
ALOGE("set_controls() can not set ctl!");
mixer_close_legacy(pMixer);
return -EINVAL;
}
}
mixer_close_legacy(pMixer);
}
return ret;
}
static void open_sound_card_policy(struct stream_out *out)
{
if (out == NULL) {
return ;
}
if (is_bitstream(out) || (is_multi_pcm(out))) {
return ;
}
for (int i = 0; i < out->num_configs; ++i) {
if (audio_is_bluetooth_out_sco_device(out->devices[i]))
return;
}
/*
* This is special process
* open all sound card to output one audio stream at the same time
*/
bool support = ((out->config.rate == 44100) || (out->config.rate == 48000));
struct audio_device *adev = out->dev;
if (support) {
out->num_configs = 0;
if(adev->dev_out[SND_OUT_SOUND_CARD_SPEAKER].card != SND_OUT_SOUND_CARD_UNKNOWN) {
out->devices[out->num_configs++] = AUDIO_DEVICE_OUT_SPEAKER;
}
if(adev->dev_out[SND_OUT_SOUND_CARD_HDMI].card != SND_OUT_SOUND_CARD_UNKNOWN) {
/*
* hdmi is taken by direct/mulit pcm output
*/
if(adev->outputs[OUTPUT_HDMI_MULTI] == NULL) {
out->devices[out->num_configs++] = AUDIO_DEVICE_OUT_AUX_DIGITAL;
}
}
if(adev->dev_out[SND_OUT_SOUND_CARD_SPDIF].card != SND_OUT_SOUND_CARD_UNKNOWN){
out->devices[out->num_configs++] = AUDIO_DEVICE_OUT_SPDIF;
}
}
}
static int open_pcm(int card, int device, int index, struct stream_out *out)
{
struct audio_device *adev = out->dev;
struct pcm_config *pcm_config = &pcm_config_ap_sco;
pcm_config->rate = adev->bt_wb_speech_enabled ? 16000 : 8000;
if (out->pcm[index] && pcm_is_ready(out->pcm[index])) {
ALOGD("pcm is ready");
return 0;
}
if (card == (int)SND_OUT_SOUND_CARD_UNKNOWN) {
ALOGE("invalid card num.");
return -EINVAL;
}
if (!out->pcm[index])
out->pcm[index] = pcm_open(card, device, PCM_OUT | PCM_MONOTONIC,
(index == (int)SND_OUT_SOUND_CARD_BT) ? pcm_config : &out->config);
if (out->pcm[index] && !pcm_is_ready(out->pcm[index])) {
ALOGE("%s failed: %s,card number = %d", __func__,
pcm_get_error(out->pcm[index]), card);
pcm_close(out->pcm[index]);
return -ENOMEM;
}
return 0;
}
/**
* @brief start_output_stream
* must be called with hw device outputs list, output stream, and hw device mutexes locked
*
* @param out
*
* @returns
*/
static int start_output_stream(struct stream_out *out)
{
struct audio_device *adev = out->dev;
int ret = 0;
int card = (int)SND_OUT_SOUND_CARD_UNKNOWN;
int device = 0;
// set defualt value to true for compatible with mid project
/*
* WORKAROUND: Do not open the same sound card with voice session's,
* because we have not supported for digital mix in libtinyalsa.
*/
if (adev->mode == AUDIO_MODE_IN_CALL) {
return 0;
}
//ALOGD("%s:%d out = %p,device = 0x%x,outputs[OUTPUT_HDMI_MULTI] = %p",__FUNCTION__,__LINE__,out,out->device,adev->outputs[OUTPUT_HDMI_MULTI]);
if (out == adev->outputs[OUTPUT_HDMI_MULTI]) {
force_non_hdmi_out_standby(adev);
} else if (adev->outputs[OUTPUT_HDMI_MULTI] &&
!adev->outputs[OUTPUT_HDMI_MULTI]->standby) {
out->disabled = true;
return 0;
}
out->disabled = false;
read_out_sound_card(out);
#ifndef SUPPORT_MULTIAUDIO
#ifdef BOX_HAL
open_sound_card_policy(out);
#endif
#endif
if (out->num_configs == 0) {
out->num_configs = 1;
out->devices[0] = out->device;
}
for (int i = 0; i < out->num_configs; ++i) {
ALOGD("%s: i = %d, device = 0x%x", __FUNCTION__, i, out->devices[i]);
if (out->devices[i] == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
audio_devices_t route_device = out->devices[i];
if (adev->owner[SOUND_CARD_HDMI] == NULL) {
card = adev->dev_out[SND_OUT_SOUND_CARD_HDMI].card;
device =adev->dev_out[SND_OUT_SOUND_CARD_HDMI].device;
#ifndef IEC958_FORAMT
#ifdef USE_DRM
// set audio mode
ret = mixer_mode_set(out);
if (ret != 0) {
ALOGE("mixer mode set error,ret=%d!",ret);
}
#endif
#endif
mixer_hdmi_set_force_bypass(out, card);
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_HDMI, out);
if (ret < 0) return ret;
if (is_bitstream(out) && ((out->config.format == PCM_FORMAT_S24_LE)
|| (out->config.format == PCM_FORMAT_IEC958_SUBFRAME_LE))) {
out->bistream = bitstream_init(out->config.format,
out->config.rate, out->config.channels);
}
if (is_multi_pcm(out) || is_bitstream(out)) {
adev->owner[SOUND_CARD_HDMI] = (int*)out;
}
}
}
if (out->devices[i] == AUDIO_DEVICE_OUT_SPEAKER ||
out->devices[i] == AUDIO_DEVICE_OUT_WIRED_HEADSET ||
out->devices[i] == AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out->devices[i] == AUDIO_DEVICE_OUT_BUS) {
audio_devices_t route_device = out->devices[i];
route_pcm_card_open(adev->dev_out[SND_OUT_SOUND_CARD_SPEAKER].card, getRouteFromDevice(route_device));
card = adev->dev_out[SND_OUT_SOUND_CARD_SPEAKER].card;
device = adev->dev_out[SND_OUT_SOUND_CARD_SPEAKER].device;
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_SPEAKER, out);
if (ret < 0) return ret;
}
if (out->devices[i] == AUDIO_DEVICE_OUT_SPDIF) {
if (adev->owner[SOUND_CARD_SPDIF] == NULL) {
card = adev->dev_out[SND_OUT_SOUND_CARD_SPDIF].card;
device = adev->dev_out[SND_OUT_SOUND_CARD_SPDIF].device;
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_SPDIF, out);
if (ret < 0) return ret;
if (is_multi_pcm(out) || is_bitstream(out)) {
adev->owner[SOUND_CARD_SPDIF] = (int*)out;
}
}
}
if (out->devices[i] == AUDIO_DEVICE_OUT_HDMI_ARC) {
audio_devices_t route_device = out->devices[i];
route_pcm_card_open(adev->dev_out[SND_OUT_SOUND_CARD_HDMI_ARC].card, getRouteFromDevice(route_device));
card = adev->dev_out[SND_OUT_SOUND_CARD_HDMI_ARC].card;
device = adev->dev_out[SND_OUT_SOUND_CARD_HDMI_ARC].device;
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_HDMI_ARC, out);
if (ret < 0) return ret;
}
#ifdef SUPPORT_VX_ROCKCHIP
if (out->devices[i] == VX_ROCKCHIP_OUT_SPDIF0) {
card = adev->dev_out[SND_OUT_SOUND_CARD_SPDIF_1].card;
device = adev->dev_out[SND_OUT_SOUND_CARD_SPDIF_1].device;
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_SPDIF_1, out);
if (ret < 0) return ret;
}
if (out->devices[i] == VX_ROCKCHIP_OUT_HDMI0) {
card = adev->dev_out[SND_OUT_SOUND_CARD_HDMI_1].card;
device = adev->dev_out[SND_OUT_SOUND_CARD_HDMI_1].device;
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_HDMI_1, out);
if (ret < 0) return ret;
}
#endif
if (audio_is_bluetooth_out_sco_device(out->devices[i])) {
card = adev->dev_out[SND_OUT_SOUND_CARD_BT].card;
device = adev->dev_out[SND_OUT_SOUND_CARD_BT].device;
struct pcm_config *pcm_config = &pcm_config_ap_sco;
pcm_config->rate = adev->bt_wb_speech_enabled?16000:8000;
ALOGD("%s pcm_open bt card number = %d, device=%d, src rate: %d dest rate:%d, wbs:%d",
__func__, card, device, out->config.rate, pcm_config->rate, adev->bt_wb_speech_enabled);
ret = open_pcm(card, device, (int)SND_OUT_SOUND_CARD_BT, out);
if (ret < 0) return ret;
ret = create_resampler(out->config.rate,
pcm_config->rate,
2,
RESAMPLER_QUALITY_DEFAULT,
NULL,
&out->resampler);
if (ret != 0) ret = -EINVAL;
}
}
adev->out_device |= out->device;
out_dump((const struct audio_stream *)out, 0);
ALOGD("%s:%d, out = %p",__FUNCTION__,__LINE__,out);
return 0;
}
/**
* @brief get_next_buffer
*
* @param buffer_provider
* @param buffer
*
* @returns
*/
static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct stream_in *in;
size_t i,size;
if (buffer_provider == NULL || buffer == NULL)
return -EINVAL;
in = (struct stream_in *)((char *)buffer_provider -
offsetof(struct stream_in, buf_provider));
if (in->pcm == NULL) {
buffer->raw = NULL;
buffer->frame_count = 0;
in->read_status = -ENODEV;
return -ENODEV;
}
if (in->frames_in == 0) {
size = pcm_frames_to_bytes(in->pcm, in->config->period_size);
in->read_status = pcm_read(in->pcm,
(void*)in->buffer, size);
if (in->read_status != 0) {
ALOGE("get_next_buffer() pcm_read error %d", in->read_status);
buffer->raw = NULL;
buffer->frame_count = 0;
return in->read_status;
}
if (in->config->channels == 2 && (!(in->device == AUDIO_DEVICE_IN_HDMI))) {
if (in->channel_flag & CH_CHECK) {
if (in->start_checkcount < SAMPLECOUNT) {
in->start_checkcount += size;
} else {
in->channel_flag = channel_check((void*)in->buffer, size / 2);
in->channel_flag &= ~CH_CHECK;
}
}
channel_fixed((void*)in->buffer, size / 2, in->channel_flag & ~CH_CHECK);
}
#ifdef RK_DENOISE_ENABLE
if ((in->mDenioseState != NULL) && !(in->device == AUDIO_DEVICE_IN_HDMI) && !(in->device == AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
rkdenoise_process(in->mDenioseState, (void*)in->buffer, size, (void*)in->buffer);
}
#endif
/*
* NOTE: We assume the input and output buffer can be overlapped
* in all effects, and the frame count is assinged to the period size
* before the resampler.
*/
audio_effect_process(&in->effects, (void *)in->buffer, (void *)in->buffer,
in->config->period_size);
//fwrite(in->buffer,pcm_frames_to_bytes(in->pcm,pcm_get_buffer_size(in->pcm)),1,in_debug);
in->frames_in = in->config->period_size;
/* Do stereo to mono conversion in place by discarding right channel */
if ((in->channel_mask == AUDIO_CHANNEL_IN_MONO)
&&(in->config->channels == 2)) {
//ALOGE("channel_mask = AUDIO_CHANNEL_IN_MONO");
for (i = 0; i < in->frames_in; i++)
in->buffer[i] = in->buffer[i * 2];
}
}
//ALOGV("pcm_frames_to_bytes(in->pcm,pcm_get_buffer_size(in->pcm)):%d",size);
buffer->frame_count = (buffer->frame_count > in->frames_in) ?
in->frames_in : buffer->frame_count;
buffer->i16 = in->buffer +
(in->config->period_size - in->frames_in) *
audio_channel_count_from_in_mask(in->channel_mask);
return in->read_status;
}
/**
* @brief release_buffer
*
* @param buffer_provider
* @param buffer
*/
static void release_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct stream_in *in;
if (buffer_provider == NULL || buffer == NULL)
return;
in = (struct stream_in *)((char *)buffer_provider -
offsetof(struct stream_in, buf_provider));
in->frames_in -= buffer->frame_count;
}
const char *device_in_sys[] = {
"hdmirx",
"rk628"
};
static bool get_hdmiin_audio_info(struct audio_device *adev, char *prop, int *value, int index)
{
char strfile[128];
FILE* file = NULL;
char info[20] = {0};
if (index >= sizeof(device_in_sys)/sizeof(device_in_sys[0]))
return false;
if (!value)
return false;
sprintf(strfile, "/sys/class/hdmirx/%s/%s", device_in_sys[index], prop);
if (access(strfile, 0)) {
ALOGD("No exist %s", strfile);
return false;
}
file = fopen(strfile, "r");
if (!file) {
ALOGD("Could reading %s property", strfile);
return false;
}
fread(info, sizeof(char), sizeof(info)/sizeof(char) - 1, file);
fclose(file);
*value = atoi(info);
return true;
}
#define STR_32KHZ "32KHZ"
#define STR_44_1KHZ "44.1KHZ"
#define STR_48KHZ "48KHZ"
/**
* @brief get_hdmiin_audio_rate
* @param
* @return hdmiin audio rate
*/
static int get_hdmiin_audio_rate(struct audio_device *adev, int index)
{
int rate;
char value[PROPERTY_VALUE_MAX] = "";
if (get_hdmiin_audio_info(adev, "audio_rate", &rate, index)) {
if (rate <= 0) {
ALOGD("%s: incorrect audio rate set to 44100", __func__);
rate = 44100;
}
return rate;
}
property_get("vendor.hdmiin.audiorate", value, STR_44_1KHZ);
if ( 0 == strncmp(value, STR_32KHZ, strlen(STR_32KHZ)) ){
rate = 32000;
} else if ( 0 == strncmp(value, STR_44_1KHZ, strlen(STR_44_1KHZ)) ){
rate = 44100;
} else if ( 0 == strncmp(value, STR_48KHZ, strlen(STR_48KHZ)) ){
rate = 48000;
} else {
rate = atoi(value);
if (rate <= 0)
rate = 44100;
}
// if hdmiin connect to codec, use 44100 sample rate
if (adev->dev_out[SND_IN_SOUND_CARD_HDMI].card
== adev->dev_out[SND_OUT_SOUND_CARD_SPEAKER].card)
rate = 44100;
return rate;
}
int create_resampler_helper(struct stream_in *in, uint32_t in_rate)
{
int ret = 0;
if (in->resampler) {
release_resampler(in->resampler);
in->resampler = NULL;
}
in->buf_provider.get_next_buffer = get_next_buffer;
in->buf_provider.release_buffer = release_buffer;
ALOGD("create resampler, channel %d, rate %d => %d",
audio_channel_count_from_in_mask(in->channel_mask),
in_rate, in->requested_rate);
ret = create_resampler(in_rate,
in->requested_rate,
audio_channel_count_from_in_mask(in->channel_mask),
RESAMPLER_QUALITY_DEFAULT,
&in->buf_provider,
&in->resampler);
if (ret != 0) {
ret = -EINVAL;
}
return ret;
}
static void config_input_effects(struct stream_in *in)
{
effect_config_t config;
int count = audio_effect_count(&in->effects);
uint16_t mask = (EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT |
EFFECT_CONFIG_ACC_MODE);
memset(&config, 0, sizeof(config));
config.inputCfg.samplingRate = in->config->rate;
config.inputCfg.channels = audio_channel_in_mask_from_count(PCM_CAPTURE_CHANNELS);
config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
config.inputCfg.mask = mask;
config.outputCfg.samplingRate = in->config->rate;
config.outputCfg.channels = audio_channel_in_mask_from_count(PCM_CAPTURE_CHANNELS);
config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
config.outputCfg.accessMode = count > 1 ?
EFFECT_BUFFER_ACCESS_ACCUMULATE :
EFFECT_BUFFER_ACCESS_WRITE;
config.outputCfg.mask = mask;
audio_effect_set_config(&in->effects, &config);
#if PCM_REFERENCE_CHANNELS
memset(&config, 0, sizeof(config));
config.inputCfg.samplingRate = in->config->rate;
config.inputCfg.channels = audio_channel_in_mask_from_count(PCM_REFERENCE_CHANNELS);
config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
config.inputCfg.mask = mask;
config.outputCfg.samplingRate = in->config->rate;
config.outputCfg.channels = audio_channel_in_mask_from_count(PCM_REFERENCE_CHANNELS);
config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
config.outputCfg.accessMode = count > 1 ?
EFFECT_BUFFER_ACCESS_ACCUMULATE :
EFFECT_BUFFER_ACCESS_WRITE;
config.outputCfg.mask = mask;
audio_effect_set_config_reverse(&in->effects, &config);
#endif
}
/**
* @brief start_input_stream
* must be called with input stream and hw device mutexes locked
*
* @param in
*
* @returns
*/
static int start_input_stream(struct stream_in *in)
{
struct audio_device *adev = in->dev;
int ret = 0;
int card = 0;
int device = 0;
int hdmiin_present = 0;
in->channel_flag = CH_CHECK;
in->start_checkcount = 0;
read_in_sound_card(in);
if (in->device == AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
in->config = &pcm_config_in_bt;
in->config->rate = adev->bt_wb_speech_enabled?16000:8000;
card = adev->dev_in[SND_IN_SOUND_CARD_BT].card;
device = adev->dev_in[SND_IN_SOUND_CARD_BT].device;
if (card != SND_IN_SOUND_CARD_UNKNOWN) {
in->pcm = pcm_open(card, device, PCM_IN | PCM_MONOTONIC, in->config);
} else {
ALOGE("%s: %d,the card number of bt is = %d",__FUNCTION__,__LINE__,card);
return -EINVAL;
}
} else if (in->device == AUDIO_DEVICE_IN_HDMI) {
char value[PROPERTY_VALUE_MAX];
in->hdmiindex = 0;
if (property_get("tvinput.hdmiin.type", value, NULL) > 0) {
ALOGD("start_input_stream tvinput.hdmiin.type=%s", value);
if(atoi(value))
in->hdmiindex = 1;
}
if (get_hdmiin_audio_info(adev, "audio_present", &hdmiin_present, in->hdmiindex)) {
if (!hdmiin_present) {
ALOGD("hdmiin audio is no present, don't open hdmiin sound");
return -EEXIST;
}
}
if (in->hdmiindex) {
card = (int)adev->dev_in[SND_IN_SOUND_CARD_HDMI_1].card;
device = adev->dev_in[SND_IN_SOUND_CARD_HDMI_1].device;
} else {
card = (int)adev->dev_in[SND_IN_SOUND_CARD_HDMI].card;
device = adev->dev_in[SND_IN_SOUND_CARD_HDMI].device;
}
if (card != SND_IN_SOUND_CARD_UNKNOWN) {
in->config->rate = get_hdmiin_audio_rate(adev, in->hdmiindex);
in->pcm = pcm_open(card, device, PCM_IN | PCM_MONOTONIC, in->config);
ALOGD("open HDMIIN %d", card);
}
} else {
ALOGD("open build mic");
in->config = &pcm_config_in;
card = adev->dev_in[SND_IN_SOUND_CARD_MIC].card;
device = adev->dev_in[SND_IN_SOUND_CARD_MIC].device;
if (card != SND_IN_SOUND_CARD_UNKNOWN) {
route_pcm_card_open(card, getRouteFromDevice(in->device | AUDIO_DEVICE_BIT_IN));
in->pcm = pcm_open(card, device, PCM_IN | PCM_MONOTONIC, in->config);
}
#ifdef RK_DENOISE_ENABLE
{
int ch = in->config->channels;
int period = in->config->period_size;
int rate = in->config->rate;
int type = 0;
{
char value[PROPERTY_VALUE_MAX];
property_get("vendor.audio.anr.speex", value, "0");
type = atoi(value);
}
if (in->mDenioseState)
rkdenoise_destroy(in->mDenioseState);
in->mDenioseState = rkdenoise_create(rate, ch, period, type ? ALG_SPX : ALG_SKV);
if (in->mDenioseState == NULL) {
ALOGW("crate rkdenoise failed!!!");
}
}
#endif
}
if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
if (in->pcm != NULL) {
ALOGE("pcm_open() failed: %s", pcm_get_error(in->pcm));
pcm_close(in->pcm);
}
ALOGD("%s open card = %d, device = %d fail", __func__, card, device);
return -ENOMEM;
}
if (in->resampler) {
release_resampler(in->resampler);
in->resampler = NULL;
}
if (in->config->rate != in->requested_rate) {
ALOGD("create resampler in->config->rate=%d in->requested_rate=%d",in->config->rate,in->requested_rate);
ret = create_resampler_helper(in, in->config->rate);
if (ret < 0 || in->resampler == NULL) {
pcm_close(in->pcm);
return -EINVAL;
}
}
/* if no supported sample rate is available, use the resampler */
if (in->resampler)
in->resampler->reset(in->resampler);
in->frames_in = 0;
adev->input_source = in->input_source;
adev->in_device = in->device;
adev->in_channel_mask = in->channel_mask;
/* initialize volume ramp */
in->ramp_frames = (CAPTURE_START_RAMP_MS * in->requested_rate) / 1000;
in->ramp_step = (uint16_t)(USHRT_MAX / in->ramp_frames);
in->ramp_vol = 0;;
/* NOTE: The following diagram shows the input stream process procedure.
*
* Audioflinger HAL
* +------------+ +------------+
* | | | |
* | Input | [yhz,ych] | resampler | [xhz,xch]
* | Thread *<==========* (optional) *<========== PCM
* | | | |
* +------------+ +------------+
*
* The audioflinger set the sampling rate and number of channels to all effects
* (yhz, ych) before creating an input stream.
*
* Because we fixed the sampling rate to 48000/44100 Hz, and the frame count
* to 5/10/16 ms, and some rockchip audio effect algorithms specify a fixed
* frame count for a sampling rate, i.e. AEC BF processing prefers 10/16ms,
* so we need to process all effects in get_next_buffer() in the case of
* resampling or non-resampling routine.
*
* And we need to set the sampling rate and number of channels before
* resampling, i.e. Set sampling rate to xhz, number of channels to xch.
*/
config_input_effects(in);
in_dump((const struct audio_stream *)in, 0);
return 0;
}
/**
* @brief get_input_buffer_size
*
* @param sample_rate
* @param format
* @param channel_count
* @param is_low_latency
*
* @returns
*/
static size_t get_input_buffer_size(unsigned int sample_rate,
audio_format_t format,
unsigned int channel_count,
bool is_low_latency)
{
const struct pcm_config *config = is_low_latency ?
&pcm_config_in_low_latency : &pcm_config_in;
size_t size;
/*
* take resampling into account and return the closest majoring
* multiple of 16 frames, as audioflinger expects audio buffers to
* be a multiple of 16 frames
*/
size = (config->period_size * sample_rate) / config->rate;
size = ((size + 15) / 16) * 16;
return size * channel_count * audio_bytes_per_sample(format);
}
/**
* @brief read_frames
* read_frames() reads frames from kernel driver, down samples to capture rate
* if necessary and output the number of frames requested to the buffer specified
*
* @param in
* @param buffer
* @param frames
*
* @returns
*/
static ssize_t read_frames(struct stream_in *in, void *buffer, ssize_t frames)
{
ssize_t frames_wr = 0;
size_t frame_size = audio_stream_in_frame_size(&in->stream);
while (frames_wr < frames) {
size_t frames_rd = frames - frames_wr;
if (in->resampler != NULL) {
in->resampler->resample_from_provider(in->resampler,
(int16_t *)((char *)buffer +
frames_wr * frame_size),
&frames_rd);
} else {
struct resampler_buffer buf = {
{
.raw = NULL,
},
.frame_count = frames_rd,
};
if (get_next_buffer(&in->buf_provider, &buf))
break;
if (buf.raw != NULL) {
memcpy((char *)buffer +
frames_wr * frame_size,
buf.raw,
buf.frame_count * frame_size);
frames_rd = buf.frame_count;
}
release_buffer(&in->buf_provider, &buf);
}
/* in->read_status is updated by getNextBuffer() also called by
* in->resampler->resample_from_provider() */
if (in->read_status != 0)
return in->read_status;
frames_wr += frames_rd;
}
return frames_wr;
}
/**
* @brief out_get_sample_rate
*
* @param stream
*
* @returns
*/
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
char value[PROPERTY_VALUE_MAX];
property_get("vendor.vts_test", value, NULL);
if (strcmp(value, "true") == 0) {
if (out->use_default_config) {
return 48000;
} else {
return out->aud_config.sample_rate;
}
} else {
return out->config.rate;
}
}
/**
* @brief out_set_sample_rate
*
* @param stream
* @param rate
*
* @returns
*/
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return -ENOSYS;
}
/**
* @brief out_get_buffer_size
*
* @param stream
*
* @returns
*/
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
return out->config.period_size *
audio_stream_out_frame_size((const struct audio_stream_out *)stream);
}
/**
* @brief out_get_channels
*
* @param stream
*
* @returns
*/
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
char value[PROPERTY_VALUE_MAX];
property_get("vendor.vts_test", value, NULL);
if (out->use_default_config) {
return AUDIO_CHANNEL_OUT_MONO;
} else {
return out->aud_config.channel_mask;
}
}
/**
* @brief out_get_format
*
* @param stream
*
* @returns
*/
static audio_format_t out_get_format(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
char value[PROPERTY_VALUE_MAX];
property_get("vendor.vts_test", value, NULL);
if (out->use_default_config) {
return AUDIO_FORMAT_PCM_16_BIT;
} else {
return out->aud_config.format;
}
}
/**
* @brief out_set_format
*
* @param stream
* @param format
*
* @returns
*/
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
return -ENOSYS;
}
/**
* @brief output_devices
* Return the set of output devices associated with active streams
* other than out. Assumes out is non-NULL and out->dev is locked.
*
* @param out
*
* @returns
*/
static audio_devices_t output_devices(struct stream_out *out)
{
struct audio_device *dev = out->dev;
enum output_type type;
audio_devices_t devices = AUDIO_DEVICE_NONE;
for (type = 0; type < OUTPUT_TOTAL; ++type) {
struct stream_out *other = dev->outputs[type];
if (other && (other != out) && !other->standby) {
// TODO no longer accurate
/* safe to access other stream without a mutex,
* because we hold the dev lock,
* which prevents the other stream from being closed
*/
devices |= other->device;
}
}
return devices;
}
/**
* @brief do_out_standby
* must be called with hw device outputs list, all out streams, and hw device mutex locked
*
* @param out
*/
static void do_out_standby(struct stream_out *out)
{
struct audio_device *adev = out->dev;
int i;
if (adev->mode == AUDIO_MODE_IN_CALL) {
if (voice_is_in_call(adev))
return;
}
ALOGD("%s,out = %p,device = 0x%x, standby = %d",__FUNCTION__,out,out->device,out->standby);
if (!out->standby) {
for (i = 0; i < SND_OUT_SOUND_CARD_MAX; i++) {
if (out->pcm[i]) {
pcm_close(out->pcm[i]);
out->pcm[i] = NULL;
}
}
out->standby = true;
out->nframes = 0;
if (out == adev->outputs[OUTPUT_HDMI_MULTI]) {
/* force standby on low latency output stream so that it can reuse HDMI driver if
* necessary when restarted */
force_non_hdmi_out_standby(adev);
}
#ifdef USE_DRM
mixer_mode_set(out);
#endif
/* re-calculate the set of active devices from other streams */
adev->out_device = output_devices(out);
#ifdef AUDIO_3A
if (adev->voice_api != NULL) {
adev->voice_api->flush();
}
#endif
route_pcm_close(PLAYBACK_OFF_ROUTE);
ALOGD("close device");
/* Skip resetting the mixer if no output device is active */
if (adev->out_device) {
route_pcm_open(getRouteFromDevice(adev->out_device));
ALOGD("change device");
}
if(adev->owner[SOUND_CARD_HDMI] == (int*)out){
adev->owner[SOUND_CARD_HDMI] = NULL;
}
if(adev->owner[SOUND_CARD_SPDIF] == (int*)out){
adev->owner[SOUND_CARD_SPDIF] = NULL;
}
bitstream_destory(&out->bistream);
}
}
/**
* @brief lock_all_outputs
* lock outputs list, all output streams, and device
*
* @param adev
*/
static void lock_all_outputs(struct audio_device *adev)
{
enum output_type type;
pthread_mutex_lock(&adev->lock_outputs);
for (type = 0; type < OUTPUT_TOTAL; ++type) {
struct stream_out *out = adev->outputs[type];
if (out)
pthread_mutex_lock(&out->lock);
}
pthread_mutex_lock(&adev->lock);
}
/**
* @brief unlock_all_outputs
* unlock device, all output streams (except specified stream), and outputs list
*
* @param adev
* @param except
*/
static void unlock_all_outputs(struct audio_device *adev, struct stream_out *except)
{
/* unlock order is irrelevant, but for cleanliness we unlock in reverse order */
pthread_mutex_unlock(&adev->lock);
enum output_type type = OUTPUT_TOTAL;
do {
struct stream_out *out = adev->outputs[--type];
if (out && out != except)
pthread_mutex_unlock(&out->lock);
} while (type != (enum output_type) 0);
pthread_mutex_unlock(&adev->lock_outputs);
}
/**
* @brief out_standby
*
* @param stream
*
* @returns
*/
static int out_standby(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
lock_all_outputs(adev);
do_out_standby(out);
unlock_all_outputs(adev, NULL);
return 0;
}
/**
* @brief out_dump
*
* @param stream
* @param fd
*
* @returns
*/
int out_dump(const struct audio_stream *stream, int fd)
{
struct stream_out *out = (struct stream_out *)stream;
for (int i = 0; i < out->num_configs; ++i) {
ALOGD("out->Device[%d] : 0x%x", i, out->devices[i]);
if (out->devices[i] == AUDIO_DEVICE_OUT_ALL_SCO) {
ALOGD("out->SampleRate : %d", pcm_config_ap_sco.rate);
ALOGD("out->Channels : %d", pcm_config_ap_sco.channels);
ALOGD("out->Format : %d", pcm_config_ap_sco.format);
ALOGD("out->PreiodSize : %d", pcm_config_ap_sco.period_size);
} else {
ALOGD("out->SampleRate : %d", out->config.rate);
ALOGD("out->Channels : %d", out->config.channels);
ALOGD("out->Format : %d", out->config.format);
ALOGD("out->PreiodSize : %d", out->config.period_size);
}
}
return 0;
}
/**
* @brief out_set_parameters
*
* @param stream
* @param kvpairs
*
* @returns
*/
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
// The set parameters here only matters when the routing devices are changed.
// When the device version is not less than 3.0, the framework will use create
// audio patch API instead of set parameters to chanage audio routing.
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
int ret;
int status = 0;
unsigned int val;
ALOGD("%s: kvpairs = %s", __func__, kvpairs);
parms = str_parms_create_str(kvpairs);
//set channel_mask
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_CHANNELS,
value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
out->aud_config.channel_mask = val;
}
// set sample rate
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_SAMPLING_RATE,
value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
out->aud_config.sample_rate = val;
}
// set format
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_FORMAT,
value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
out->aud_config.format = val;
}
str_parms_destroy(parms);
ALOGV("%s: exit: status(%d)", __func__, status);
return status;
}
/*
* function: get support formats
* Query supported formats. The response is a '|' separated list of strings from audio_format_t enum
* e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT"
*/
static int stream_get_parameter_formats(const struct audio_stream *stream,
struct str_parms *query,
struct str_parms *reply)
{
struct stream_out *out = (struct stream_out *)stream;
int avail = 1024;
char value[avail];
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
memset(value,0,avail);
// set support pcm 16 bit default
strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
return 0;
}
return -1;
}
/*
* function: get support channels
* Query supported channel masks. The response is a '|' separated list of strings from
* audio_channel_mask_t enum
* e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO"
*/
static int stream_get_parameter_channels(struct str_parms *query,
struct str_parms *reply,
audio_channel_mask_t *supported_channel_masks)
{
char value[1024];
size_t i, j;
bool first = true;
if(str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)){
value[0] = '\0';
i = 0;
/* the last entry in supported_channel_masks[] is always 0 */
while (supported_channel_masks[i] != 0) {
for (j = 0; j < ARRAY_SIZE(channels_name_to_enum_table); j++) {
if (channels_name_to_enum_table[j].value == supported_channel_masks[i]) {
if (!first) {
strcat(value, "|");
}
strcat(value, channels_name_to_enum_table[j].name);
first = false;
break;
}
}
i++;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
return 0;
}
return -1;
}
/*
* function: get support sample_rates
* Query supported sampling rates. The response is a '|' separated list of integer values
* e.g: ""sup_sampling_rates=44100|48000"
*/
static int stream_get_parameter_rates(struct str_parms *query,
struct str_parms *reply,
uint32_t *supported_sample_rates)
{
char value[256];
int ret = -1;
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
value[0] = '\0';
int cursor = 0;
int i = 0;
while(supported_sample_rates[i]){
int avail = sizeof(value) - cursor;
ret = snprintf(value + cursor, avail, "%s%d",
cursor > 0 ? "|" : "",
supported_sample_rates[i]);
if (ret < 0 || ret > avail){
value[cursor] = '\0';
break;
}
cursor += ret;
++i;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value);
return 0;
}
return -1;
}
/**
* @brief out_get_parameters
*
* @param stream
* @param keys
*
* @returns
*/
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
ALOGD("%s: keys = %s", __func__, keys);
struct stream_out *out = (struct stream_out *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str = NULL;
struct str_parms *reply = str_parms_create();
out->use_default_config = true;
if (stream_get_parameter_formats(stream,query,reply) == 0) {
str = str_parms_to_str(reply);
} else if (stream_get_parameter_channels(query, reply, &out->supported_channel_masks[0]) == 0) {
str = str_parms_to_str(reply);
} else if (stream_get_parameter_rates(query, reply, &out->supported_sample_rates[0]) == 0) {
str = str_parms_to_str(reply);
} else {
ALOGD("%s,str_parms_get_str failed !",__func__);
str = strdup("");
}
str_parms_destroy(query);
str_parms_destroy(reply);
ALOGV("%s,exit -- str = %s",__func__,str);
return str;
}
/**
* @brief out_get_latency
*
* @param stream
*
* @returns
*/
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
struct stream_out *out = (struct stream_out *)stream;
return (out->config.period_size * out->config.period_count * 1000) /
out->config.rate;
}
/**
* @brief out_set_volume
*
* @param stream
* @param left
* @param right
*
* @returns
*/
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
out->volume[0] = left;
out->volume[1] = right;
ALOGD("vol:%f", left);
/* The mutex lock is not needed, because the client
* is not allowed to close the stream concurrently with this API
* pthread_mutex_lock(&adev->lock_outputs);
*/
bool is_HDMI = out == adev->outputs[OUTPUT_HDMI_MULTI];
/* pthread_mutex_unlock(&adev->lock_outputs); */
if (is_HDMI) {
/* only take left channel into account: the API is for stereo anyway */
out->muted = (left == 0.0f);
return 0;
}
return -ENOSYS;
}
/**
* @brief dump_out_data
*
* @param buffer bytes
*/
static void dump_out_data(const void* buffer,size_t bytes)
{
char value[PROPERTY_VALUE_MAX];
property_get("vendor.audio.record", value, "0");
int size = atoi(value);
if (size <= 0)
return ;
ALOGD("dump pcm file.");
static FILE* fd = NULL;
static int offset = 0;
if (fd == NULL) {
fd=fopen("/data/misc/audioserver/debug.pcm","wb+");
if(fd == NULL) {
ALOGD("DEBUG open /data/debug.pcm ,errno = %s",strerror(errno));
offset = 0;
}
}
if (fd != NULL) {
fwrite(buffer,bytes,1,fd);
offset += bytes;
fflush(fd);
if(offset >= size*1024*1024) {
fclose(fd);
fd = NULL;
offset = 0;
property_set("vendor.audio.record", "0");
ALOGD("TEST playback pcmfile end");
}
}
}
static void dump_in_data(const void* buffer, size_t bytes)
{
static int offset = 0;
static FILE* fd = NULL;
char value[PROPERTY_VALUE_MAX];
property_get("vendor.audio.record.in", value, "0");
int size = atoi(value);
if (size > 0) {
if(fd == NULL) {
fd=fopen("/data/misc/audioserver/debug_in.pcm","wb+");
if(fd == NULL) {
ALOGD("DEBUG open /data/misc/audioserver/debug_in.pcm ,errno = %s",strerror(errno));
} else {
ALOGD("dump pcm to file /data/misc/audioserver/debug_in.pcm");
}
offset = 0;
}
}
if (fd != NULL) {
ALOGD("dump in pcm %zu bytes", bytes);
fwrite(buffer,bytes,1,fd);
offset += bytes;
fflush(fd);
if (offset >= size*1024*1024) {
fclose(fd);
fd = NULL;
offset = 0;
property_set("vendor.audio.record.in", "0");
ALOGD("TEST record pcmfile end");
}
}
}
static void check_hdmi_reconnect(struct stream_out *out)
{
if (out == NULL) {
return ;
}
struct audio_device *adev = out->dev;
lock_all_outputs(adev);
/*
* if snd_reopen is set to true, this means we need to reopen sound card.
* There are a situation, we need to do this:
* current stream is bistream over hdmi, and hdmi is unpluged and plug later,
* the driver of hdmi may init the hdmi in pcm mode automatically, according the
* implement of driver of hdmi. If we contiune send bitstream to hdmi open in pcm mode,
* hdmi may make noies or mute.
*/
if (out->snd_reopen && !out->standby)
{
/*
* standby sound cards
* the driver of hdmi will auto init with last configurations,
* so, we don't need close and reopen sound card of hdmi here.
* If driver of hdmi not config the hdmi with last output configurations,
* please open this codes to close and reopen sound card of hdmi.
*/
// do_out_standby(out);
// reset_bitstream_buf(out);
}
unlock_all_outputs(adev,NULL);
/*
* audio hal recived the msg of hdmi plugin, and other part of sdk will reviced it too.
* Other part(maybe hwc) will config hdmi after it reviced the msg.
* Audio must wait other part(maybe hwc) codes config hdmi finish, before send bitstream datas to hdmi
*/
if (out->snd_reopen && is_bitstream(out) && is_type_in_outdevices(out, AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
#ifdef USE_DRM
const char* PATH = "/sys/class/drm/card0-HDMI-A-1/enabled";
#else
const char* PATH = "/sys/class/display/HDMI/enabled";
#endif
if (access(PATH, R_OK) != 0) {
/*
* in most test, the time is 700~800ms between received msg of hdmi plug in
* and hdmi init finish, so we sleep 1 sec here if no way to get the status of hdmi.
*/
usleep(1000000);
} else {
/*
* read this node to judge the status of hdmi is config finish?
*/
char buffer[1024];
int counter = 200;
FILE* file = NULL;
while (counter >= 0 && ((file = fopen(PATH,"r")) != NULL)) {
int size = fread(buffer,1,sizeof(buffer),file);
if(size >= 0) {
if(strstr(buffer,"enabled")) {
fclose(file);
usleep(10000);
break;
}
}
usleep(10000);
counter --;
fclose(file);
}
}
ALOGD("%s: out = %p",__FUNCTION__,out);
out->snd_reopen = false;
}
}
static void out_mute_data(struct stream_out *out,void* buffer,size_t bytes)
{
struct audio_device *adev = out->dev;
bool mute = false;
#ifdef MUTE_WHEN_SCREEN_OFF
mute = adev->screenOff;
#endif
// for some special customer
char value[PROPERTY_VALUE_MAX];
property_get("vendor.audio.mute", value, "false");
if (!strcasecmp(value,"true")) {
mute = true;
}
if (out->muted || mute){
memset((void *)buffer, 0, bytes);
}
}
static int bitstream_write_data(struct stream_out *out, void* buffer, size_t bytes)
{
if ((out == NULL) || (buffer == NULL) || (bytes <= 0)) {
ALOGD("%s: %d, input parameter is invalid",__FUNCTION__,__LINE__);
return -1;
}
struct audio_device *adev = out->dev;
int ret = 0;
if (is_type_in_outdevices(out, AUDIO_DEVICE_OUT_AUX_DIGITAL) && (is_multi_pcm(out) || is_bitstream(out))) {
int card = adev->dev_out[SND_OUT_SOUND_CARD_HDMI].card;
if ((card != SND_OUT_SOUND_CARD_UNKNOWN) && (out->pcm[SND_OUT_SOUND_CARD_HDMI] != NULL)) {
if(out->config.format == PCM_FORMAT_S16_LE){
out_mute_data(out,buffer,bytes);
dump_out_data(buffer, bytes);
ret = pcm_write(out->pcm[SND_OUT_SOUND_CARD_HDMI], (void *)buffer, bytes);
} else if(out->config.format == PCM_FORMAT_S24_LE ||
out->config.format == PCM_FORMAT_IEC958_SUBFRAME_LE){
char *outBuffer = NULL;
int outSize = 0;
ret = bitstream_encode(out->bistream, (char*)buffer, (int)bytes, &outBuffer, &outSize);
if (ret == 0 && outSize > 0) {
out_mute_data(out,(void*)outBuffer, outSize);
dump_out_data((void*)outBuffer, outSize);
ret = pcm_write(out->pcm[SND_OUT_SOUND_CARD_HDMI], (void *)outBuffer, outSize);
if (ret != 0) {
ALOGD("%s:%d pcm_write error, out = %p, errno = %d, %s",
__FUNCTION__, __LINE__, out, errno,
pcm_get_error(out->pcm[SND_OUT_SOUND_CARD_HDMI]));
}
}
}
} else {
ALOGD("%s: %d: HDMI sound card not open",__FUNCTION__,__LINE__);
ret = -1;
}
}
return ret;
}
/*
* process volume of non-fix device. for exp: spk
*
*/
static void out_pcm_volume_process(struct stream_out *out, void *buffer, size_t len) {
if ((out == NULL) || (buffer == NULL) || (len <= 0)) {
return ;
}
int16_t *pcm = (int16_t *)buffer;
float left = out->volume[0];
int channel = out->config.channels;
int frames = len / audio_stream_out_frame_size((const struct audio_stream_out *)out);
for (int frame = 0; frame < frames; frame ++) {
int32_t temp = 0;
for (int ch = 0; ch < channel; ch++) {
temp = (int32_t)(pcm[ch] * left);
if (temp > 0x7fff) {
temp = 0x7fff;
} else if (temp < -0x8000) {
temp = -0x8000;
}
pcm[ch] = temp;
}
pcm += channel;
}
}
/*
* process volume of one multi pcm frame
* The multi pcm output no using mixer,so the can't control by volume setting,
* so here we process multi pcm datas with volume value.
*
*/
static void out_multi_pcm_volume_process(struct stream_out *out, void *buffer)
{
if ((out == NULL) || (buffer == NULL)){
return ;
}
int format = out->config.format;
int channel = out->config.channels;
if (format == PCM_FORMAT_S16_LE) {
float left = out->volume[0];
short *pcm = (short*)buffer;
float temp = 0;
for (int ch = 0; ch < channel; ch++) {
temp = (float)pcm[ch];
pcm[ch] = (short)(temp*left);
}
}
}
/*
* switch LFE and FC of one multi pcm frame
* swtich Front Center's datas and Low Frequency datas
* 5.1 FL+FR+FC+LFE+BL+BR
* 5.1(side) FL+FR+FC+LFE+SL+SR
* 7.1 FL+FR+FC+LFE+SL+SR+BL+BR
* the datas needed in HDMI is:
* FL+FR+LFE+FC+SL+SR+BL+BR
*/
static void out_multi_pcm_switch_fc_lfe(struct stream_out *out, void *buffer)
{
if ((out == NULL) || (buffer == NULL)) {
return ;
}
const int CENTER = 2;
const int LFE = 3;
int channel = out->config.channels;
int format = out->config.format;
audio_channel_mask_t channel_mask = out->channel_mask;
bool hasLFE = ((channel_mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) != 0);
if (format == PCM_FORMAT_S16_LE) {
short *pcm = (short*)buffer;
short temp = 0;
if (hasLFE && ((channel == 6) || (channel == 8))) {
// Front Center's datas
temp = pcm[CENTER];
// swap FC and Low Frequency Effect's datas
pcm[CENTER] = pcm[LFE];
pcm[LFE] = temp;
}
}
}
static void out_multi_pcm_process(struct stream_out *out, const void *buffer, size_t len) {
if((out == NULL) || (buffer == NULL) || (len <= 0)){
return ;
}
int format = out->config.format;
// only process PCM16
if (format == PCM_FORMAT_S16_LE) {
short *pcm = (short*)buffer;
int channel = out->config.channels;
int frames = len / audio_stream_out_frame_size((const struct audio_stream_out *)out);
for (int frame = 0; frame < frames; frame ++){
out_multi_pcm_volume_process(out, pcm);
out_multi_pcm_switch_fc_lfe(out, pcm);
pcm += channel;
}
}
}
/**
* @brief out_write
*
* @param stream
* @param buffer
* @param bytes
*
* @returns
*/
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret = 0;
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
size_t newbytes = bytes * 2;
int i,card;
uint64_t startTime = 0, endTime, cost;
if (adev->mode == AUDIO_MODE_IN_CALL) {
if (voice_is_in_call(adev))
return bytes;
}
/* FIXME This comment is no longer correct
* acquiring hw device mutex systematically is useful if a low
* priority thread is waiting on the output stream mutex - e.g.
* executing out_set_parameters() while holding the hw device
* mutex
*/
//check_hdmi_reconnect(out);
pthread_mutex_lock(&out->lock);
if (out->standby) {
pthread_mutex_unlock(&out->lock);
lock_all_outputs(adev);
if (!out->standby) {
unlock_all_outputs(adev, out);
goto false_alarm;
}
ret = start_output_stream(out);
if (ret < 0) {
unlock_all_outputs(adev, NULL);
goto final_exit;
}
out->standby = false;
unlock_all_outputs(adev, out);
}
false_alarm:
if (out->disabled) {
ret = -EPIPE;
goto exit;
}
#ifdef AUDIO_3A
if (adev->voice_api != NULL) {
int ret = 0;
adev->voice_api->queuePlaybackBuffer(buffer, bytes);
ret = adev->voice_api->getPlaybackBuffer(buffer, bytes);
if (ret < 0) {
memset((char *)buffer, 0x00, bytes);
}
}
#endif
/* Write to all active PCMs */
if (is_type_in_outdevices(out, AUDIO_DEVICE_OUT_AUX_DIGITAL) && is_bitstream(out)) {
ret = bitstream_write_data(out, (void*)buffer, bytes);
if(ret < 0) {
goto exit;
}
} else {
if(is_multi_pcm(out)) {
if(out->device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
out_multi_pcm_process(out, buffer, bytes);
}
} else {
/*
* NOTE: We have not supported effects for bitstream and multi-channel
* (more than 2 channels) stream of HDMI/SPDIF.
*
* We assume the input and output buffer can be overlapped
* in all effects, and the frame count is assinged in get_buffer_size().
*/
audio_effect_process(&out->effects, (void *)buffer, (void *)buffer,
out->config.period_size);
}
out_mute_data(out,(void*)buffer,bytes);
dump_out_data(buffer, bytes);
ret = -1;
for (i = 0; i < SND_OUT_SOUND_CARD_MAX; i++)
if (out->pcm[i]) {
if (i == SND_OUT_SOUND_CARD_BT) {
// HARD CODE FIXME 48000 stereo -> 8000 stereo
size_t inFrameCount = bytes/2/2;
int destRate = out->config.rate;
int srcRate = adev->bt_wb_speech_enabled? 16000:8000;
int coefficient = (destRate/srcRate);
size_t outFrameCount = inFrameCount/coefficient;
int16_t out_buffer[outFrameCount*2];
memset(out_buffer, 0x00, outFrameCount*2);
out->resampler->resample_from_input(out->resampler,
(int16_t *)buffer,
&inFrameCount,
out_buffer,
&outFrameCount);
ret = pcm_write(out->pcm[i], (void *)out_buffer, outFrameCount*2*2);
if (ret != 0)
break;
#ifdef SET_VOL_IN_HAL
} else if (i == SND_OUT_SOUND_CARD_SPEAKER) {
short out_buffer[bytes];
memcpy(out_buffer, buffer, bytes);
out_pcm_volume_process(out, out_buffer, bytes);
ret = pcm_write(out->pcm[i], (void *)out_buffer, bytes);
if (ret != 0) {
ALOGD("%s:%d pcm_write error, errno = %d, %s",
__FUNCTION__, __LINE__, errno, pcm_get_error(out->pcm[i]));
break;
}
#endif
} else {
/*
* do not write hdmi/spdif snd sound if they are taken by other bitstream/multi channle pcm stream
*/
if(((i == SND_OUT_SOUND_CARD_HDMI) && (adev->owner[SOUND_CARD_HDMI] != (int*)out) && (adev->owner[SOUND_CARD_HDMI] != NULL)) ||
((i == SND_OUT_SOUND_CARD_SPDIF) && (adev->owner[SOUND_CARD_SPDIF] != (int*)out) && (adev->owner[SOUND_CARD_SPDIF] != NULL))){
continue;
}
ret = pcm_write(out->pcm[i], (void *)buffer, bytes);
if (ret != 0) {
ALOGD("%s:%d pcm_write error, errno = %d, %s",
__FUNCTION__, __LINE__, errno, pcm_get_error(out->pcm[i]));
break;
}
}
}
}
exit:
pthread_mutex_unlock(&out->lock);
final_exit:
{
// For PCM we always consume the buffer and return #bytes regardless of ret.
out->written += bytes / (out->config.channels * sizeof(short));
out->nframes = out->written;
}
if (ret != 0) {
ALOGV("AudioData write error , keep slience! ret = %d", ret);
endTime = getRelativeUs();
cost = endTime - startTime;
uint64_t time = (uint64_t)bytes * 1000000ll / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common);
if (cost < time) {
usleep((int)(time - cost));
}
/*
* HDR video will set hdmi to hdr mode in kernel, this operation will
* lead hdmi card enter stop state(see sound/soc/codecs/hdmi-codec.c),
* and pcm_write will alway fail, and pcm_prepare in pcm_write can not
* recovery this state. It can only recovery by reopen sound card.
*/
if (!out->standby && !out->disabled) {
lock_all_outputs(adev);
do_out_standby(out);
unlock_all_outputs(adev, NULL);
}
}
return bytes;
}
/**
* @brief out_get_render_position
*
* @param stream
* @param dsp_frames
*
* @returns
*/
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
struct stream_out *out = (struct stream_out *)stream;
*dsp_frames = out->nframes;
return 0;
}
/**
* @brief out_add_audio_effect
*
* @param stream
* @param effect
*
* @returns
*/
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGD("%s: effect %p", __func__, effect);
pthread_mutex_lock(&out->lock);
audio_effect_add(&out->effects, effect);
pthread_mutex_unlock(&out->lock);
return 0;
}
/**
* @brief out_remove_audio_effect
*
* @param stream
* @param effect
*
* @returns
*/
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGD("%s: effect %p", __func__, effect);
pthread_mutex_lock(&out->lock);
audio_effect_remove(&out->effects, effect);
pthread_mutex_unlock(&out->lock);
return 0;
}
/**
* @brief out_get_next_write_timestamp
*
* @param stream
* @param timestamp
*
* @returns
*/
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
ALOGV("%s: %d Entered", __FUNCTION__, __LINE__);
return -ENOSYS;
}
/**
* @brief out_get_presentation_position
*
* @param stream
* @param frames
* @param timestamp
*
* @returns
*/
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
struct stream_out *out = (struct stream_out *)stream;
int ret = -1;
pthread_mutex_lock(&out->lock);
int i;
// There is a question how to implement this correctly when there is more than one PCM stream.
// We are just interested in the frames pending for playback in the kernel buffer here,
// not the total played since start. The current behavior should be safe because the
// cases where both cards are active are marginal.
for (i = 0; i < SND_OUT_SOUND_CARD_MAX; i++)
if (out->pcm[i]) {
unsigned int avail;
//ALOGD("===============%s,%d==============",__FUNCTION__,__LINE__);
if (pcm_get_htimestamp(out->pcm[i], &avail, timestamp) == 0) {
size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
//ALOGD("===============%s,%d==============",__FUNCTION__,__LINE__);
// FIXME This calculation is incorrect if there is buffering after app processor
int64_t signed_frames = out->written - kernel_buffer_size + avail;
//signed_frames -= 17;
//ALOGV("============singed_frames:%lld=======",signed_frames);
//ALOGV("============timestamp:%lld==========",timestamp);
// It would be unusual for this value to be negative, but check just in case ...
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
break;
}
}
pthread_mutex_unlock(&out->lock);
return ret;
}
/**
* @brief in_get_sample_rate
* audio_stream_in implementation
*
* @param stream
*
* @returns
*/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
//ALOGV("%s:get requested_rate : %d ",__FUNCTION__,in->requested_rate);
return in->requested_rate;
}
/**
* @brief in_set_sample_rate
*
* @param stream
* @param rate
*
* @returns
*/
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return 0;
}
/**
* @brief in_get_channels
*
* @param stream
*
* @returns
*/
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
//ALOGV("%s:get channel_mask : %d ",__FUNCTION__,in->channel_mask);
return in->channel_mask;
}
/**
* @brief in_get_buffer_size
*
* @param stream
*
* @returns
*/
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
return get_input_buffer_size(in->requested_rate,
AUDIO_FORMAT_PCM_16_BIT,
audio_channel_count_from_in_mask(in_get_channels(stream)),
(in->flags & AUDIO_INPUT_FLAG_FAST) != 0);
}
/**
* @brief in_get_format
*
* @param stream
*
* @returns
*/
static audio_format_t in_get_format(const struct audio_stream *stream)
{
return AUDIO_FORMAT_PCM_16_BIT;
}
/**
* @brief in_set_format
*
* @param stream
* @param format
*
* @returns
*/
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
return -ENOSYS;
}
/**
* @brief do_in_standby
* must be called with in stream and hw device mutex locked
*
* @param in
*/
static void do_in_standby(struct stream_in *in)
{
struct audio_device *adev = in->dev;
if (!in->standby) {
pcm_close(in->pcm);
in->pcm = NULL;
if (in->device == AUDIO_DEVICE_IN_HDMI) {
route_pcm_close(HDMI_IN_CAPTURE_OFF_ROUTE);
}
in->dev->input_source = AUDIO_SOURCE_DEFAULT;
in->dev->in_device = AUDIO_DEVICE_NONE;
in->dev->in_channel_mask = 0;
in->standby = true;
route_pcm_close(CAPTURE_OFF_ROUTE);
}
}
/**
* @brief in_standby
*
* @param stream
*
* @returns
*/
static int in_standby(struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
pthread_mutex_lock(&in->lock);
pthread_mutex_lock(&in->dev->lock);
do_in_standby(in);
pthread_mutex_unlock(&in->dev->lock);
pthread_mutex_unlock(&in->lock);
return 0;
}
/**
* @brief in_dump
*
* @param stream
* @param fd
*
* @returns
*/
int in_dump(const struct audio_stream *stream, int fd)
{
struct stream_in *in = (struct stream_in *)stream;
ALOGD("in->Device : 0x%x", in->device);
ALOGD("in->SampleRate : %d", in->config->rate);
ALOGD("in->Channels : %d", in->config->channels);
ALOGD("in->Formate : %d", in->config->format);
ALOGD("in->PreiodSize : %d", in->config->period_size);
return 0;
}
/**
* @brief in_set_parameters
*
* @param stream
* @param kvpairs
*
* @returns
*/
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
struct str_parms *parms;
char value[32];
int ret;
int status = 0;
unsigned int val;
bool apply_now = false;
ALOGV("%s: kvpairs = %s", __func__, kvpairs);
parms = str_parms_create_str(kvpairs);
//set channel_mask
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_CHANNELS,
value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
in->channel_mask = val;
}
// set sample rate
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_SAMPLING_RATE,
value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
in->requested_rate = val;
}
pthread_mutex_lock(&in->lock);
pthread_mutex_lock(&adev->lock);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE,
value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
/* no audio source uses val == 0 */
if ((in->input_source != val) && (val != 0)) {
in->input_source = val;
apply_now = !in->standby;
}
}
if (apply_now) {
adev->input_source = in->input_source;
adev->in_device = in->device;
route_pcm_open(getRouteFromDevice(in->device | AUDIO_DEVICE_BIT_IN));
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
str_parms_destroy(parms);
ALOGV("%s: exit: status(%d)", __func__, status);
return status;
}
/**
* @brief in_get_parameters
*
* @param stream
* @param keys
*
* @returns
*/
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
ALOGD("%s: keys = %s", __func__, keys);
struct stream_in *in = (struct stream_in *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str = NULL;
struct str_parms *reply = str_parms_create();
if (stream_get_parameter_formats(stream,query,reply) == 0) {
str = str_parms_to_str(reply);
} else if (stream_get_parameter_channels(query, reply, &in->supported_channel_masks[0]) == 0) {
str = str_parms_to_str(reply);
} else if (stream_get_parameter_rates(query, reply, &in->supported_sample_rates[0]) == 0) {
str = str_parms_to_str(reply);
} else {
ALOGD("%s,str_parms_get_str failed !",__func__);
str = strdup("");
}
str_parms_destroy(query);
str_parms_destroy(reply);
ALOGV("%s,exit -- str = %s",__func__,str);
return str;
}
/**
* @brief in_set_gain
*
* @param stream
* @param gain
*
* @returns
*/
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
/**
* @brief in_apply_ramp
*
* @param in
* @param buffer
* @param frames
*/
static void in_apply_ramp(struct stream_in *in, int16_t *buffer, size_t frames)
{
size_t i;
uint16_t vol = in->ramp_vol;
uint16_t step = in->ramp_step;
frames = (frames < in->ramp_frames) ? frames : in->ramp_frames;
if (in->channel_mask == AUDIO_CHANNEL_IN_MONO)
for (i = 0; i < frames; i++) {
buffer[i] = (int16_t)((buffer[i] * vol) >> 16);
vol += step;
}
else
for (i = 0; i < frames; i++) {
buffer[2*i] = (int16_t)((buffer[2*i] * vol) >> 16);
buffer[2*i + 1] = (int16_t)((buffer[2*i + 1] * vol) >> 16);
vol += step;
}
in->ramp_vol = vol;
in->ramp_frames -= frames;
}
/**
* @brief in_read
*
* @param stream
* @param buffer
* @param bytes
*
* @returns
*/
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
int ret = 0;
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
size_t frames_rq = bytes / audio_stream_in_frame_size(stream);
size_t frames_rd = 0;
if (in->device == AUDIO_DEVICE_IN_HDMI) {
unsigned int rate = get_hdmiin_audio_rate(adev, in->hdmiindex);
if(rate != in->config->rate){
ALOGD("HDMI-In: rate is changed: %d -> %d, restart input stream",
in->config->rate, rate);
pthread_mutex_lock(&in->lock);
do_in_standby(in);
pthread_mutex_unlock(&in->lock);
}
}
/*
* acquiring hw device mutex systematically is useful if a low
* priority thread is waiting on the input stream mutex - e.g.
* executing in_set_parameters() while holding the hw device
* mutex
*/
pthread_mutex_lock(&in->lock);
if (in->standby) {
pthread_mutex_lock(&adev->lock);
ret = start_input_stream(in);
pthread_mutex_unlock(&adev->lock);
if (ret < 0)
goto exit;
in->standby = false;
#ifdef AUDIO_3A
if (adev->voice_api != NULL) {
adev->voice_api->start();
}
#endif
}
/*if (in->num_preprocessors != 0)
ret = process_frames(in, buffer, frames_rq);
else */
//ALOGV("%s:frames_rq:%d",__FUNCTION__,frames_rq);
frames_rd = read_frames(in, buffer, frames_rq);
if (in->read_status) {
ret = -EPIPE;
goto exit;
} else if (frames_rd > 0) {
in->frames_read += frames_rd;
bytes = frames_rd * audio_stream_in_frame_size(stream);
}
dump_in_data(buffer, bytes);
#ifdef AUDIO_3A
do {
if (adev->voice_api != NULL) {
int ret = 0;
ret = adev->voice_api->quueCaputureBuffer(buffer, bytes);
if (ret < 0) break;
ret = adev->voice_api->getCapureBuffer(buffer, bytes);
if (ret < 0) memset(buffer, 0x00, bytes);
}
} while (0);
#endif
if (in->ramp_frames > 0)
in_apply_ramp(in, buffer, frames_rq);
/*
* Instead of writing zeroes here, we could trust the hardware
* to always provide zeroes when muted.
*/
if (ret == 0 && adev->mic_mute)
memset(buffer, 0, bytes);
#ifdef ALSA_IN_DEBUG
fwrite(buffer, bytes, 1, in_debug);
#endif
exit:
if (ret < 0) {
memset(buffer, 0, bytes);
usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&stream->common));
do_in_standby(in);
}
pthread_mutex_unlock(&in->lock);
return bytes;
}
/**
* @brief in_get_input_frames_lost
*
* @param stream
*
* @returns
*/
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
/**
* @brief in_add_audio_effect
*
* @param stream
* @param effect
*
* @returns
*/
static int in_add_audio_effect(const struct audio_stream *stream,
effect_handle_t effect)
{
struct stream_in *in = (struct stream_in *)stream;
ALOGD("%s: effect %p", __func__, effect);
pthread_mutex_lock(&in->lock);
audio_effect_add(&in->effects, effect);
pthread_mutex_unlock(&in->lock);
return 0;
}
/**
* @brief in_remove_audio_effect
*
* @param stream
* @param effect
*
* @returns
*/
static int in_remove_audio_effect(const struct audio_stream *stream,
effect_handle_t effect)
{
struct stream_in *in = (struct stream_in *)stream;
ALOGD("%s: effect %p", __func__, effect);
pthread_mutex_lock(&in->lock);
audio_effect_remove(&in->effects, effect);
pthread_mutex_unlock(&in->lock);
return 0;
}
static int adev_get_microphones(const struct audio_hw_device *dev,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count)
{
struct audio_device *adev = (struct audio_device *)dev;
size_t actual_mic_count = 0;
int card_no = 0;
char snd_card_node_id[100]={0};
char snd_card_node_cap[100]={0};
char address[32] = "bottom";
do{
sprintf(snd_card_node_id, "/proc/asound/card%d/id", card_no);
if (access(snd_card_node_id,F_OK) == -1) break;
sprintf(snd_card_node_cap, "/proc/asound/card%d/pcm0c/info", card_no);
if (access(snd_card_node_cap,F_OK) == -1) continue;
actual_mic_count++;
}while(++card_no);
mic_array->device = AUDIO_DEVICE_IN_BUILTIN_MIC;
strcpy(mic_array->address,address);
ALOGD("%s,get capture mic actual_mic_count =%zu", __func__, actual_mic_count);
*mic_count = actual_mic_count;
return 0;
}
static int in_get_capture_position(const struct audio_stream_in *stream,
int64_t *frames, int64_t *time)
{
if (stream == NULL || frames == NULL || time == NULL) {
return -EINVAL;
}
struct stream_in *in = (struct stream_in *)stream;
int ret = -ENOSYS;
pthread_mutex_lock(&in->lock);
// note: ST sessions do not close the alsa pcm driver synchronously
// on standby. Therefore, we may return an error even though the
// pcm stream is still opened.
if (in->standby) {
ALOGD("skip when standby is true.");
goto exit;
}
if (in->pcm) {
struct timespec timestamp;
unsigned int avail;
if (pcm_get_htimestamp(in->pcm, &avail, ×tamp) == 0) {
*frames = in->frames_read + avail;
*time = timestamp.tv_sec * 1000000000LL + timestamp.tv_nsec;
ret = 0;
ALOGV("Pos: %" PRId64 " %" PRId64, *time, *frames);
}
}
exit:
pthread_mutex_unlock(&in->lock);
return ret;
}
static int in_get_active_microphones(const struct audio_stream_in *stream,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
pthread_mutex_lock(&in->lock);
pthread_mutex_lock(&adev->lock);
size_t actual_mic_count = 0;
int card_no = 0;
char snd_card_node_id[100]={0};
char snd_card_node_cap[100]={0};
char snd_card_info[100]={0};
char snd_card_state[255]={0};
do{
sprintf(snd_card_node_id, "/proc/asound/card%d/id", card_no);
if (access(snd_card_node_id,F_OK) == -1) break;
sprintf(snd_card_node_cap, "/proc/asound/card%d/pcm0c/info", card_no);
if (access(snd_card_node_cap,F_OK) == -1) {
continue;
} else {
sprintf(snd_card_info, "/proc/asound/card%d/pcm0c/sub0/status", card_no);
int fd;
fd = open(snd_card_info, O_RDONLY);
if (fd < 0) {
ALOGE("%s,failed to open node: %s", __func__, snd_card_info);
} else {
int length = read(fd, snd_card_state, sizeof(snd_card_state) -1);
snd_card_state[length] = 0;
if (strcmp(snd_card_state, "closed") != 0) actual_mic_count++;
}
close(fd);
}
} while(++card_no);
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
ALOGD("%s,get active mic actual_mic_count =%zu", __func__, actual_mic_count);
*mic_count = actual_mic_count;
return 0;
}
/*
* get support channels mask of hdmi from parsing edid of hdmi
*/
static void get_hdmi_support_channels_masks(struct stream_out *out)
{
if(out == NULL)
return ;
int channels = get_hdmi_audio_speaker_allocation(&out->hdmi_audio);
switch (channels) {
case AUDIO_CHANNEL_OUT_5POINT1:
ALOGD("%s: HDMI Support 5.1 channels pcm",__FUNCTION__);
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_STEREO;
break;
case AUDIO_CHANNEL_OUT_7POINT1:
ALOGD("%s: HDMI Support 7.1 channels pcm",__FUNCTION__);
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
break;
case AUDIO_CHANNEL_OUT_STEREO:
default:
ALOGD("%s: HDMI Support 2 channels pcm",__FUNCTION__);
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_MONO;
break;
}
}
/**
* @brief adev_open_output_stream
*
* @param dev
* @param handle
* @param devices
* @param flags
* @param config
* @param stream_out
* @param __unused
*
* @returns
*/
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address __unused)
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
int ret;
enum output_type type = OUTPUT_LOW_LATENCY;
bool isPcm = audio_is_linear_pcm(config->format);
bool isBitstream = false;
ALOGD("audio hal adev_open_output_stream devices = 0x%x, flags = %d, samplerate = %d,format = 0x%x",
devices, flags, config->sample_rate,config->format);
// only support PCM or IEC61937 format
if (!audio_has_proportional_frames(config->format)) {
ALOGD("%s: format = 0x%x not support", __FUNCTION__, config->format);
return -1;
}
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
if (!out)
return -ENOMEM;
if ((flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) ||
(config->format == AUDIO_FORMAT_IEC61937)) {
isBitstream = true;
}
/*get default supported channel_mask*/
memset(out->supported_channel_masks, 0, sizeof(out->supported_channel_masks));
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_MONO;
/*get default supported sample_rate*/
memset(out->supported_sample_rates, 0, sizeof(out->supported_sample_rates));
out->supported_sample_rates[0] = 44100;
out->supported_sample_rates[1] = 48000;
if(config != NULL)
memcpy(&(out->aud_config),config,sizeof(struct audio_config));
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
if (devices == AUDIO_DEVICE_NONE)
devices = AUDIO_DEVICE_OUT_SPEAKER;
out->device = devices;
/*
* set output_direct_mode to LPCM, means data is not multi pcm or bitstream datas.
* set output_direct to false, means data is 2 channels pcm
*/
out->output_direct_mode = LPCM;
out->output_direct = false;
out->snd_reopen = false;
out->use_default_config = false;
out->volume[0] = out->volume[1] = 1.0f;
out->bistream = NULL;
init_hdmi_audio(&out->hdmi_audio);
if(devices == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
parse_hdmi_audio(&out->hdmi_audio);
get_hdmi_support_channels_masks(out);
}
if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
if (isBitstream) {
ALOGD("%s:out = %p HDMI Bitstream",__FUNCTION__,out);
out->channel_mask = config->channel_mask;
if (isValidSamplerate(config->sample_rate)) {
out->config = pcm_config_direct;
out->config.rate = config->sample_rate;
out->output_direct = true;
int channel = audio_channel_count_from_out_mask(config->channel_mask);
if (channel == 8 && config->sample_rate == 192000) {
out->output_direct_mode = HBR;
} else {
out->output_direct_mode = NLPCM;
}
if (out->config.format == PCM_FORMAT_S24_LE) {
if (config->sample_rate >= 176400) {
out->config.period_size = 1024 * 4;
} else {
out->config.period_size = 2048;
}
} else {
out->config.period_size = config->sample_rate/100; // 10ms
}
#ifdef RK3128 // only 3128 using 16bit to bitstream
out->config.format = PCM_FORMAT_S16_LE;
#endif
type = OUTPUT_HDMI_MULTI;
} else {
out->config = pcm_config;
out->config.rate = 44100;
ALOGE("hdmi bitstream samplerate %d unsupport", config->sample_rate);
}
out->config.channels = audio_channel_count_from_out_mask(config->channel_mask);
if (out->config.channels < 2)
out->config.channels = 2;
out->pcm_device = PCM_DEVICE;
out->device = AUDIO_DEVICE_OUT_AUX_DIGITAL;
} else if (isPcm){ // multi pcm
if (config->sample_rate == 0)
config->sample_rate = HDMI_MULTI_DEFAULT_SAMPLING_RATE;
if (config->channel_mask == 0)
config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
int layout = get_hdmi_audio_speaker_allocation(&out->hdmi_audio);
unsigned int mask = (layout&config->channel_mask);
ALOGD("%s:out = %p HDMI multi pcm: layout = 0x%x,mask = 0x%x",
__FUNCTION__,out,layout,mask);
// current hdmi allocation(speaker) only support MONO or STEREO
if(mask <= (int)AUDIO_CHANNEL_OUT_STEREO) {
ALOGD("%s:out = %p input stream is multi pcm,channle mask = 0x%x,but hdmi not support,mixer it to stereo output",
__FUNCTION__,out,config->channel_mask);
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
out->config = pcm_config;
out->pcm_device = PCM_DEVICE;
type = OUTPUT_LOW_LATENCY;
out->device = AUDIO_DEVICE_OUT_AUX_DIGITAL;
out->output_direct = false;
} else {
/*
* maybe input audio stream is 7.1 channels,
* but hdmi only support 5.1, we also output 7.1 for default.
* Is better than output 2 channels after mixer?
* If customer like output 2 channles data after mixer,
* modify codes here
*/
out->channel_mask = config->channel_mask;
out->config = pcm_config_hdmi_multi;
out->config.rate = config->sample_rate;
out->config.channels = audio_channel_count_from_out_mask(config->channel_mask);
out->pcm_device = PCM_DEVICE;
type = OUTPUT_HDMI_MULTI;
out->device = AUDIO_DEVICE_OUT_AUX_DIGITAL;
out->output_direct = true;
}
} else {
ALOGD("Not any bitstream mode!");
}
} else if ((devices & AUDIO_DEVICE_OUT_SPDIF) && isBitstream) {
ALOGD("%s:out = %p Spdif Bitstream",__FUNCTION__,out);
out->channel_mask = config->channel_mask;
out->config = pcm_config_direct;
if ((config->sample_rate == 48000) ||
(config->sample_rate == 32000) ||
(config->sample_rate == 44100)) {
out->config.rate = config->sample_rate;
out->config.format = PCM_FORMAT_S16_LE;
out->config.period_size = config->sample_rate/100; // 10ms
} else {
out->config.rate = 44100;
ALOGE("spdif passthrough samplerate %d is unsupport",config->sample_rate);
}
out->config.channels = audio_channel_count_from_out_mask(config->channel_mask);
devices = AUDIO_DEVICE_OUT_SPDIF;
out->pcm_device = PCM_DEVICE;
out->output_direct = true;
type = OUTPUT_HDMI_MULTI;
out->device = AUDIO_DEVICE_OUT_SPDIF;
out->output_direct_mode = NLPCM;
} else {
out->config = pcm_config;
out->pcm_device = PCM_DEVICE;
type = OUTPUT_LOW_LATENCY;
}
} else if (flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
out->config = pcm_config_deep;
out->pcm_device = PCM_DEVICE_DEEP;
type = OUTPUT_DEEP_BUF;
} else {
out->config = pcm_config;
out->pcm_device = PCM_DEVICE;
type = OUTPUT_LOW_LATENCY;
}
ALOGD("out->config.rate = %d, out->config.channels = %d out->config.format = %d",
out->config.rate, out->config.channels, out->config.format);
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.get_latency = out_get_latency;
out->stream.set_volume = out_set_volume;
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
out->handle = handle;
out->dev = adev;
out->standby = true;
out->nframes = 0;
pthread_mutex_lock(&adev->lock_outputs);
if (adev->outputs[type]) {
pthread_mutex_unlock(&adev->lock_outputs);
ret = -EBUSY;
goto err_open;
}
adev->outputs[type] = out;
pthread_mutex_unlock(&adev->lock_outputs);
adev_add_stream_to_list(out->dev, &out->dev->output_stream_list, &out->list_node);
list_init(&out->effects);
*stream_out = &out->stream;
return 0;
err_open:
if (out != NULL) {
destory_hdmi_audio(&out->hdmi_audio);
free(out);
}
*stream_out = NULL;
return ret;
}
/**
* @brief adev_close_output_stream
*
* @param dev
* @param stream
*/
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
struct audio_device *adev;
enum output_type type;
ALOGD("adev_close_output_stream!");
out_standby(&stream->common);
adev = (struct audio_device *)dev;
pthread_mutex_lock(&adev->lock_outputs);
for (type = 0; type < OUTPUT_TOTAL; ++type) {
if (adev->outputs[type] == (struct stream_out *) stream) {
adev->outputs[type] = NULL;
break;
}
}
{
struct stream_out *out = (struct stream_out *)stream;
device_lock(out->dev);
list_remove(&out->list_node);
device_unlock(out->dev);
destory_hdmi_audio(&out->hdmi_audio);
}
pthread_mutex_unlock(&adev->lock_outputs);
free(stream);
}
/**
* @brief adev_set_parameters
*
* @param dev
* @param kvpairs
*
* @returns
*/
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
struct audio_device *adev = (struct audio_device *)dev;
struct str_parms *parms = NULL;
char value[32] = "";
/*
* ret is the result of str_parms_get_str,
* if no paramter which str_parms_get_str to get, it will return result < 0 always.
* For example: kvpairs = connect=1024 is coming
* str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SCREEN_STATE,value, sizeof(value))
* will return result < 0,this means no screen_state in parms
*/
int ret = 0;
/*
* status is the result of one process,
* For example: kvpairs = screen_state=on is coming,
* str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SCREEN_STATE,value, sizeof(value))
* will return result >= 0,this means screen is on, we can do something,
* if the things we do is correct, we set status = 0, or status < 0 means fail.
*/
int status = 0;
ALOGD("%s: kvpairs = %s", __func__, kvpairs);
parms = str_parms_create_str(kvpairs);
pthread_mutex_lock(&adev->lock);
// screen state off/on
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SCREEN_STATE, // screen_state
value, sizeof(value));
if (ret >= 0) {
if(strcmp(value,"on") == 0){
adev->screenOff = false;
} else if(strcmp(value,"off") == 0){
adev->screenOff = true;
}
}
#ifdef AUDIO_BITSTREAM_REOPEN_HDMI
// hdmi reconnect
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, // hdmi reconnect
value, sizeof(value));
if (ret >= 0) {
int device = atoi(value);
if(device == (int)AUDIO_DEVICE_OUT_AUX_DIGITAL){
struct stream_out *out = adev->outputs[OUTPUT_HDMI_MULTI];
if((out != NULL) && is_bitstream(out) && (out->device == AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
ALOGD("%s: hdmi connect when audio stream is output over hdmi, do something,out = %p",__FUNCTION__,out);
out->snd_reopen = true;
}
}
}
#endif
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value));
if (ret >= 0) {
adev->bt_wb_speech_enabled = !strcmp(value, AUDIO_PARAMETER_VALUE_ON);
ALOGD("%s: adev:0x%p, bt_wb_speech_enabled = %d",
__func__, adev, adev->bt_wb_speech_enabled);
}
ret = str_parms_get_str(parms, "BT_SCO", value, sizeof(value));
if (ret >= 0) {
adev->bt_sco_reroute ^= !strcmp(value, AUDIO_PARAMETER_VALUE_ON);
if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) != 0){
adev->bt_wb_speech_enabled = false;
}
}
ALOGD("%s:bt_wb_speech_enabled = %d, sco reroute=%u",
__func__, adev->bt_wb_speech_enabled, adev->bt_sco_reroute);
pthread_mutex_unlock(&adev->lock);
str_parms_destroy(parms);
return status;
}
/*
* get support formats for bitstream
* There is no stand interface in andorid to get the formats can be bistream,
* so we extend get parameter to report formats
*/
static int get_support_bitstream_formats(struct str_parms *query,
struct str_parms *reply)
{
int avail = 1024;
char value[avail];
struct hdmi_audio_infors hdmi_edid;
init_hdmi_audio(&hdmi_edid);
const char* AUDIO_PARAMETER_STREAM_SUP_BITSTREAM_FORMAT = "sup_bitstream_formats";
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_BITSTREAM_FORMAT)) {
memset(value,0,avail);
// get the format can be bistream?
if(parse_hdmi_audio(&hdmi_edid) >= 0){
int cursor = 0;
for(int i = 0; i < ARRAY_SIZE(sSurroundFormat); i++){
if(is_support_format(&hdmi_edid,sSurroundFormat[i].format)){
avail -= cursor;
int length = snprintf(value + cursor, avail, "%s%s",
cursor > 0 ? "|" : "",
sSurroundFormat[i].value);
if (length < 0 || length >= avail) {
break;
}
cursor += length;
}
}
}
destory_hdmi_audio(&hdmi_edid);
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_BITSTREAM_FORMAT, value);
return 0;
}
return -1;
}
/**
* @brief adev_get_parameters
*
* @param dev
* @param keys
*
* @returns
*/
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
struct audio_device *adev = (struct audio_device *)dev;
struct str_parms *parms = str_parms_create_str(keys);
struct str_parms *reply = str_parms_create();
char *str = NULL;
ALOGD("%s: keys = %s",__FUNCTION__,keys);
if (str_parms_has_key(parms, "ec_supported")) {
str_parms_destroy(parms);
parms = str_parms_create_str("ec_supported=yes");
str = str_parms_to_str(parms);
}
#ifdef PRIMARY_HAL
else if (str_parms_has_key(parms, "hdmi0_registered")) {
str_parms_destroy(parms);
if (check_out_card_registered(HDMI_OUT_NAME)) {
parms = str_parms_create_str("hdmi0_registered=1");
} else {
parms = str_parms_create_str("hdmi0_registered=0");
}
str = str_parms_to_str(parms);
} else if (str_parms_has_key(parms, "hdmi1_registered")) {
str_parms_destroy(parms);
if (check_out_card_registered(HDMI_1_OUT_NAME)) {
parms = str_parms_create_str("hdmi1_registered=1");
} else {
parms = str_parms_create_str("hdmi1_registered=0");
}
str = str_parms_to_str(parms);
} else if (str_parms_has_key(parms, "dp0_registered")) {
str_parms_destroy(parms);
if (check_out_card_registered(SPDIF_OUT_NAME)) {
parms = str_parms_create_str("dp0_registered=1");
} else {
parms = str_parms_create_str("dp0_registered=0");
}
str = str_parms_to_str(parms);
} else if (str_parms_has_key(parms, "dp1_registered")) {
str_parms_destroy(parms);
if (check_out_card_registered(SPDIF_1_OUT_NAME)) {
parms = str_parms_create_str("dp1_registered=1");
} else {
parms = str_parms_create_str("dp1_registered=0");
}
str = str_parms_to_str(parms);
}
#endif
else if (get_support_bitstream_formats(parms,reply) == 0) {
str = str_parms_to_str(reply);
} else {
str = strdup("");
}
str_parms_destroy(parms);
str_parms_destroy(reply);
return str;
}
/**
* @brief adev_init_check
*
* @param dev
*
* @returns
*/
static int adev_init_check(const struct audio_hw_device *dev)
{
return 0;
}
/**
* @brief adev_set_voice_volume
*
* @param dev
* @param volume
*
* @returns
*/
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
int ret = 0;
struct audio_device *adev = (struct audio_device *)dev;
voice_set_volume(adev, volume);
if(adev->mode == AUDIO_MODE_IN_CALL) {
if (volume < 0.0) {
volume = 0.0;
} else if (volume > 1.0) {
volume = 1.0;
}
const char *mixer_ctl_name = "Speaker Playback Volume";
ret = route_set_voice_volume(mixer_ctl_name,volume);
}
return ret;
}
/**
* @brief adev_set_master_volume
*
* @param dev
* @param volume
*
* @returns
*/
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
return -ENOSYS;
}
/**
* @brief adev_set_mode
*
* @param dev
* @param mode
*
* @returns
*/
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
struct audio_device *adev = (struct audio_device *)dev;
ALOGD("%s: set_mode = %d", __func__, mode);
if (adev->mode != mode) {
if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION)) {
adev->voice.route = AUDIO_DEVICE_NONE;
if (voice_is_in_call(adev)) {
voice_stop_incall(adev);
}
}
adev->mode = mode;
}
return 0;
}
/**
* @brief adev_set_mic_mute
*
* @param dev
* @param state
*
* @returns
*/
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
struct audio_device *adev = (struct audio_device *)dev;
adev->mic_mute = state;
if (adev->mode == AUDIO_MODE_IN_CALL) {
voice_set_mic_mute(adev, state);
}
return 0;
}
/**
* @brief adev_get_mic_mute
*
* @param dev
* @param state
*
* @returns
*/
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
struct audio_device *adev = (struct audio_device *)dev;
*state = adev->mic_mute;
return 0;
}
/**
* @brief adev_get_input_buffer_size
*
* @param dev
* @param config
*
* @returns
*/
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
return get_input_buffer_size(config->sample_rate, config->format,
audio_channel_count_from_in_mask(config->channel_mask),
false /* is_low_latency: since we don't know, be conservative */);
}
/**
* @brief adev_open_input_stream
*
* @param dev
* @param handle
* @param devices
* @param config
* @param stream_in
* @param flags
* @param __unused
* @param __unused
*
* @returns
*/
static int adev_open_input_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
audio_input_flags_t flags,
const char *address __unused,
audio_source_t source __unused)
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_in *in;
int ret;
ALOGD("audio hal adev_open_input_stream devices = 0x%x, flags = %d, config->samplerate = %d,config->channel_mask = %x",
devices, flags, config->sample_rate,config->channel_mask);
*stream_in = NULL;
#ifdef ALSA_IN_DEBUG
in_debug = fopen("/data/debug.pcm","wb");//please touch /data/debug.pcm first
#endif
/* Respond with a request for mono if a different format is given. */
//ALOGV("%s:config->channel_mask %d",__FUNCTION__,config->channel_mask);
if (/*config->channel_mask != AUDIO_CHANNEL_IN_MONO &&
config->channel_mask != AUDIO_CHANNEL_IN_FRONT_BACK*/
config->channel_mask != AUDIO_CHANNEL_IN_STEREO) {
config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
ALOGE("%s:channel is not support",__FUNCTION__);
return -EINVAL;
}
if (config->sample_rate == 0 ) {
config->sample_rate = 44100;
ALOGW("%s: rate is not support",__FUNCTION__);
}
in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
if (!in)
return -ENOMEM;
/*get default supported channel_mask*/
memset(in->supported_channel_masks, 0, sizeof(in->supported_channel_masks));
in->supported_channel_masks[0] = AUDIO_CHANNEL_IN_STEREO;
in->supported_channel_masks[1] = AUDIO_CHANNEL_IN_MONO;
/*get default supported sample_rate*/
memset(in->supported_sample_rates, 0, sizeof(in->supported_sample_rates));
in->supported_sample_rates[0] = 44100;
in->supported_sample_rates[1] = 48000;
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
in->stream.common.set_format = in_set_format;
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.common.add_audio_effect = in_add_audio_effect;
in->stream.common.remove_audio_effect = in_remove_audio_effect;
in->stream.set_gain = in_set_gain;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
in->stream.get_active_microphones = in_get_active_microphones;
in->stream.get_capture_position = in_get_capture_position;
#ifdef RK_DENOISE_ENABLE
in->mDenioseState = NULL;
#endif
in->dev = adev;
in->standby = true;
in->requested_rate = config->sample_rate;
in->input_source = AUDIO_SOURCE_DEFAULT;
in->device = devices;
in->handle = handle;
in->io_handle = handle;
in->channel_mask = config->channel_mask;
in->resampler = NULL;
if (in->device == AUDIO_DEVICE_IN_HDMI) {
ALOGD("HDMI-In: use low latency");
flags |= AUDIO_INPUT_FLAG_FAST;
}
in->flags = flags;
struct pcm_config *pcm_config = flags & AUDIO_INPUT_FLAG_FAST ?
&pcm_config_in_low_latency : &pcm_config_in;
if (in->device == AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
pcm_config = &pcm_config_in_bt;
pcm_config->rate = adev->bt_wb_speech_enabled? 16000:8000;
}
in->config = pcm_config;
in->buffer = malloc(pcm_config->period_size * pcm_config->channels
* audio_stream_in_frame_size(&in->stream));
if (!in->buffer) {
ret = -ENOMEM;
goto err_malloc;
}
if (in->device == AUDIO_DEVICE_IN_HDMI) {
goto out;
}
#ifdef AUDIO_3A
ALOGD("voice process has opened, try to create voice process!");
adev->voice_api = rk_voiceprocess_create(DEFAULT_PLAYBACK_SAMPLERATE,
DEFAULT_PLAYBACK_CHANNELS,
in->requested_rate,
audio_channel_count_from_in_mask(in->channel_mask));
if (adev->voice_api == NULL) {
ALOGE("crate voice process failed!");
}
#endif
out:
adev_add_stream_to_list(in->dev, &in->dev->input_stream_list, &in->list_node);
list_init(&in->effects);
*stream_in = &in->stream;
return 0;
err_resampler:
free(in->buffer);
err_malloc:
free(in);
return ret;
}
/**
* @brief adev_close_input_stream
*
* @param dev
* @param stream
*/
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = (struct audio_device *)dev;
ALOGD("%s",__FUNCTION__);
device_lock(in->dev);
list_remove(&in->list_node);
device_unlock(in->dev);
in_standby(&stream->common);
if (in->resampler) {
release_resampler(in->resampler);
in->resampler = NULL;
}
#ifdef ALSA_IN_DEBUG
fclose(in_debug);
#endif
#ifdef AUDIO_3A
if (adev->voice_api != NULL) {
rk_voiceprocess_destory();
adev->voice_api = NULL;
}
#endif
#ifdef RK_DENOISE_ENABLE
if (in->mDenioseState)
rkdenoise_destroy(in->mDenioseState);
in->mDenioseState = NULL;
#endif
free(in->buffer);
free(stream);
}
/**
* @brief adev_dump
*
* @param device
* @param fd
*
* @returns
*/
static int adev_dump(const audio_hw_device_t *device, int fd)
{
return 0;
}
/**
* @brief adev_close
*
* @param device
*
* @returns
*/
static int adev_close(hw_device_t *device)
{
struct audio_device *adev = (struct audio_device *)device;
//audio_route_free(adev->ar);
route_uninit();
free(device);
return 0;
}
static int adev_create_audio_patch(struct audio_hw_device *dev,
unsigned int num_sources,
const struct audio_port_config *sources,
unsigned int num_sinks,
const struct audio_port_config *sinks,
audio_patch_handle_t *handle)
{
if (num_sources != 1 || num_sinks == 0 || num_sinks > AUDIO_PATCH_PORTS_MAX) {
ALOGE("%s unsupport num sources:%d or sinks:%d", __func__, num_sources, num_sinks);
return -EINVAL;
}
ALOGD("%s num_sources:%d, num_sinks:%d, %s(%x)->%s(%x), handle:%p"
, __func__, num_sources, num_sinks
, (sources[0].type == AUDIO_PORT_TYPE_MIX) ? "mix" : "device"
, (sources[0].type == AUDIO_PORT_TYPE_MIX) ?
sources[0].ext.mix.handle : sources[0].ext.device.type
, (sinks[0].type == AUDIO_PORT_TYPE_MIX) ? "mix" : "device"
, (sinks[0].type == AUDIO_PORT_TYPE_MIX) ?
sinks[0].ext.mix.handle : sinks[0].ext.device.type
, handle);
struct audio_device *adev = (struct audio_device *)dev;
const struct audio_port_config *src_port_config = sources;
const struct audio_port_config *sink_port_config = sinks;
bool generatedPatchHandle = false;
if (*handle == AUDIO_PATCH_HANDLE_NONE) {
*handle = ++adev->next_patch_handle;
generatedPatchHandle = true;
}
struct stream_in *in = NULL;
struct stream_out *out = NULL;
audio_io_handle_t io_handle = 0;
audio_patch_handle_t *patch_handle = NULL;
bool wasStandby = true;
int ret = 0;
/**
* 1. device --> mix (recording)
* 2. mix --> device (playback)
*/
if (sources[0].type == AUDIO_PORT_TYPE_DEVICE
&& sinks[0].type == AUDIO_PORT_TYPE_MIX) {
if (num_sinks != 1) {
return -EINVAL;
}
ALOGD("%s recording", __func__);
device_lock(adev);
in = adev_get_stream_in_by_io_handle_l(adev, sinks[0].ext.mix.handle);
if (in == NULL) {
ALOGE("%s()can not find stream with handle(%d)", __func__, sinks[0].ext.mix.handle);
device_unlock(adev);
return -EINVAL;
}
io_handle = in->handle;
patch_handle = &in->patch_handle;
wasStandby = in->standby;
if (!generatedPatchHandle && *patch_handle != *handle) {
ALOGE("%s() the patch handle(%d) does not match recorded one(%d) for stream "
"with handle(%d) when creating audio patch",
__func__, *handle, *patch_handle, io_handle);
device_unlock(adev);
return -EINVAL;
}
in->device = sources[0].ext.device.type;
device_unlock(adev);
pthread_mutex_lock(&in->lock);
device_lock(adev);
do_in_standby(in);
device_unlock(adev);
*patch_handle = *handle;
if (!wasStandby) {
device_lock(adev);
if (in != NULL) {
ret = start_input_stream(in);
if (ret < 0) {
ALOGE("%s() start_input_stream fail", __func__);
device_unlock(adev);
pthread_mutex_unlock(&in->lock);
return -EINVAL;
}
in->standby = false;
}
device_unlock(adev);
}
pthread_mutex_unlock(&in->lock);
} else if (sources[0].type == AUDIO_PORT_TYPE_MIX
&& sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
for (unsigned int i = 0; i < num_sinks; i++) {
if (sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
ALOGE("%s() invalid sink type %#x for mix source", __func__, sinks[i].type);
return -EINVAL;
}
}
ALOGD("%s playback address:%s", __func__, sinks[0].ext.device.address);
device_lock(adev);
out = adev_get_stream_out_by_io_handle_l(adev, sources[0].ext.mix.handle);
if (out == NULL) {
ALOGE("%s()can not find stream with handle(%d)", __func__, sinks[0].ext.mix.handle);
device_unlock(adev);
return -EINVAL;
}
io_handle = out->handle;
patch_handle = &out->patch_handle;
wasStandby = out->standby;
if (!generatedPatchHandle && *patch_handle != *handle) {
ALOGE("%s() the patch handle(%d) does not match recorded one(%d) for stream "
"with handle(%d) when creating audio patch",
__func__, *handle, *patch_handle, io_handle);
device_unlock(adev);
return -EINVAL;
}
out->num_configs = num_sinks;
for (unsigned int i = 0; i < num_sinks; ++i) {
out->devices[i] = sinks[i].ext.device.type;
}
device_unlock(adev);
lock_all_outputs(adev);
do_out_standby(out);
unlock_all_outputs(adev, NULL);
*patch_handle = *handle;
if (!wasStandby) {
lock_all_outputs(adev);
if (in != NULL) {
ret = start_output_stream(out);
if (ret < 0) {
ALOGE("%s() start_output_stream fail", __func__);
unlock_all_outputs(adev, NULL);
return -EINVAL;
}
out->standby = false;
}
unlock_all_outputs(adev, NULL);
}
/* FIXME: We only support one route for voice call. */
if (adev->mode == AUDIO_MODE_IN_CALL) {
if (adev->voice.route != out->devices[0]) {
if (voice_is_in_call(adev))
voice_stop_incall(adev);
adev->voice.route = out->devices[0];
voice_start_incall(adev);
}
}
} else {
// All other cases are invalid.
ALOGE("%s() invalid case", __func__);
return -EINVAL;
}
return 0;
}
static int adev_release_audio_patch(struct audio_hw_device *dev,
audio_patch_handle_t handle)
{
struct audio_device* adev = (struct audio_device*) dev;
ALOGD("%s in", __func__);
device_lock(adev);
struct stream_out *out = adev_get_stream_out_by_patch_handle_l(adev, handle);
device_unlock(adev);
if (out != NULL) {
lock_all_outputs(adev);
do_out_standby(out);
out->patch_handle = AUDIO_PATCH_HANDLE_NONE;
unlock_all_outputs(adev, NULL);
return 0;
}
device_lock(adev);
struct stream_in *in = adev_get_stream_in_by_patch_handle_l(adev, handle);
device_unlock(adev);
if (in != NULL) {
pthread_mutex_lock(&in->lock);
lock_all_outputs(adev);
do_in_standby(in);
unlock_all_outputs(adev, NULL);
in->patch_handle = AUDIO_PATCH_HANDLE_NONE;
pthread_mutex_unlock(&in->lock);
return 0;
}
ALOGE("%s cannot find stream with patch handle as %d", __func__, handle);
return -EINVAL;
}
static int adev_get_audio_port(struct audio_hw_device *dev,
struct audio_port *port)
{
return -EINVAL;
}
static int adev_set_audio_port_config(struct audio_hw_device *dev,
const struct audio_port_config *config)
{
return -EINVAL;
}
static void adev_open_init(struct audio_device *adev)
{
ALOGD("%s",__func__);
int i = 0;
adev->mic_mute = false;
adev->screenOff = false;
#ifdef AUDIO_3A
adev->voice_api = NULL;
#endif
adev->input_source = AUDIO_SOURCE_DEFAULT;
for(i =0; i < OUTPUT_TOTAL; i++){
adev->outputs[i] = NULL;
}
set_default_dev_info(adev->dev_out, SND_OUT_SOUND_CARD_MAX, 1);
set_default_dev_info(adev->dev_in, SND_IN_SOUND_CARD_MAX, 1);
adev->dev_out[SND_OUT_SOUND_CARD_SPEAKER].id = "SPEAKER";
adev->dev_out[SND_OUT_SOUND_CARD_HDMI].id = "HDMI";
adev->dev_out[SND_OUT_SOUND_CARD_HDMI_1].id = "HDMI_1";
adev->dev_out[SND_OUT_SOUND_CARD_SPDIF].id = "SPDIF";
adev->dev_out[SND_OUT_SOUND_CARD_SPDIF_1].id = "SPDIF_1";
adev->dev_out[SND_OUT_SOUND_CARD_BT].id = "BT";
adev->dev_out[SND_OUT_SOUND_CARD_HDMI_ARC].id = "HDMI_ARC";
adev->dev_in[SND_IN_SOUND_CARD_MIC].id = "MIC";
adev->dev_in[SND_IN_SOUND_CARD_BT].id = "BT";
adev->owner[0] = NULL;
adev->owner[1] = NULL;
char value[PROPERTY_VALUE_MAX];
if (property_get("vendor.audio.period_size", value, NULL) > 0) {
pcm_config.period_size = atoi(value);
pcm_config_in.period_size = pcm_config.period_size;
}
if (property_get("vendor.audio.in_period_size", value, NULL) > 0)
pcm_config_in.period_size = atoi(value);
}
/**
* @brief adev_open
*
* @param module
* @param name
* @param device
*
* @returns
*/
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct audio_device *adev;
int ret;
ALOGD(AUDIO_HAL_VERSION);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = calloc(1, sizeof(struct audio_device));
if (!adev)
return -ENOMEM;
list_init(&adev->output_stream_list);
list_init(&adev->input_stream_list);
adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_3_0;
adev->hw_device.common.module = (struct hw_module_t *) module;
adev->hw_device.common.close = adev_close;
adev->hw_device.init_check = adev_init_check;
adev->hw_device.set_voice_volume = adev_set_voice_volume;
adev->hw_device.set_master_volume = adev_set_master_volume;
adev->hw_device.set_mode = adev_set_mode;
adev->hw_device.set_mic_mute = adev_set_mic_mute;
adev->hw_device.get_mic_mute = adev_get_mic_mute;
adev->hw_device.set_parameters = adev_set_parameters;
adev->hw_device.get_parameters = adev_get_parameters;
adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
adev->hw_device.open_output_stream = adev_open_output_stream;
adev->hw_device.close_output_stream = adev_close_output_stream;
adev->hw_device.open_input_stream = adev_open_input_stream;
adev->hw_device.close_input_stream = adev_close_input_stream;
adev->hw_device.dump = adev_dump;
adev->hw_device.get_microphones = adev_get_microphones;
//add for version 3.0
adev->hw_device.create_audio_patch = adev_create_audio_patch;
adev->hw_device.release_audio_patch = adev_release_audio_patch;
adev->hw_device.get_audio_port = adev_get_audio_port;
adev->hw_device.set_audio_port_config = adev_set_audio_port_config;
//adev->ar = audio_route_init(MIXER_CARD, NULL);
//route_init();
/* adev->cur_route_id initial value is 0 and such that first device
* selection is always applied by select_devices() */
*device = &adev->hw_device.common;
adev->bt_wb_speech_enabled = false;
adev->bt_sco_reroute = 0;
voice_init(adev);
adev_open_init(adev);
return 0;
}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
struct audio_module HAL_MODULE_INFO_SYM = {
.common = {
.tag = HARDWARE_MODULE_TAG,
.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
.hal_api_version = HARDWARE_HAL_API_VERSION,
.id = AUDIO_HARDWARE_MODULE_ID,
.name = "Manta audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};
1、使用【最接近】Rockchip的原厂Android14，KickPi 7的SDK。
DTS适应EVB1编译，没有声音。
2、直接使用 飞凌OK3576-C开发板 的DTS。
Y:\OK3576-android-source\kernel-6.1\arch\arm64\boot\dts\rockchip\OK3576-C-android.dts
还是没有声音。
1|console:/sdcard #
1|console:/sdcard # cat /proc/asound/cards
0 [rockchipnau8822]: rockchip-nau882 - rockchip-nau8822
rockchip-nau8822
1 [rockchipdp0 ]: rockchip-dp0 - rockchip-dp0
rockchip-dp0
2 [rockchiphdmi ]: rockchip-hdmi - rockchip-hdmi
rockchip-hdmi
console:/sdcard #
console:/sdcard # ls -l /dev/snd/
total 0
crw-rw---- 1 system audio 116, 4 2025-07-27 15:17 controlC0
crw-rw---- 1 system audio 116, 6 2025-07-27 15:17 controlC1
crw-rw---- 1 system audio 116, 9 2025-07-27 15:17 controlC2
crw-rw---- 1 system audio 116, 3 2025-07-27 15:17 pcmC0D0c
crw-rw---- 1 system audio 116, 2 2025-07-27 15:17 pcmC0D0p
crw-rw---- 1 system audio 116, 5 2025-07-27 15:17 pcmC1D0p
crw-rw---- 1 system audio 116, 8 2025-07-27 15:17 pcmC2D0c
crw-rw---- 1 system audio 116, 7 2025-07-27 15:17 pcmC2D0p
crw-rw---- 1 system audio 116, 33 2025-07-27 15:17 timer
console:/sdcard #
console:/sdcard # ll
total 52115
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Alarms
drwxrws--x 5 media_rw media_rw 3452 2025-07-27 15:17 Android
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Audiobooks
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 DCIM
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Documents
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Download
drwxrws--- 3 u0_a74 media_rw 3452 2025-07-27 15:17 Movies
drwxrws--- 3 u0_a74 media_rw 3452 2025-07-27 15:17 Music
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Notifications
drwxrws--- 3 u0_a74 media_rw 3452 2025-07-27 15:17 Pictures
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Podcasts
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Recordings
drwxrws--- 2 u0_a74 media_rw 3452 2025-07-27 15:17 Ringtones
-rw-rw---- 1 u0_a74 media_rw 53268598 2025-07-27 12:45 kiss8.wav
console:/sdcard # tin
tinycap tinymix tinypcminfo tinyplay
console:/sdcard # tinyp
tinypcminfo tinyplay
console:/sdcard # tinyplay
Usage: tinyplay file.wav [-D card] [-d device] [-p period_size] [-n n_periods]
1|console:/sdcard #
1|console:/sdcard # tin
tinycap tinymix tinypcminfo tinyplay
1|console:/sdcard # tinyplay kiss8.wav -D 0 -d 0
Playing sample: 2 ch, 44100 hz, 16 bit 53268480 bytes
3、那就分析/比对内核/driver。
直接查找nau88c22*.c，找不到！
很容易/根据经验/艰难得知驱动名是：nau8822.c
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217$
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217$ cd kernel-6.1/
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$ find . -name nau88c22*.c
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$ find . -name nau8822*.c
./sound/soc/codecs/nau8822 - 副本.c
./sound/soc/codecs/nau8822.c
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$
rootroot@rootroot-HP-ZHAN-66-Pro-A-14-G3:~/rk3576-android14.0-20250217/kernel-6.1$
https://blog.youkuaiyun.com/qq_39104123/article/details/115752708?spm=1001.2101.3001.6650.2&utm_medium=distribute.pc_relevant.none-task-blog-2%7Edefault%7EBlogCommendFromBaidu%7EPaidSort-2-115752708-blog-143410504.235%5Ev43%5Epc_blog_bottom_relevance_base8&depth_1-utm_source=distribute.pc_relevant.none-task-blog-2%7Edefault%7EBlogCommendFromBaidu%7EPaidSort-2-115752708-blog-143410504.235%5Ev43%5Epc_blog_bottom_relevance_base8&utm_relevant_index=4
基于imx8mq的NAU8822音频驱动开发
https://blog.youkuaiyun.com/qq_39104123/article/details/115752708
优快云：NAU88C22YG 驱动名：nau8822.c
4、和 飞凌OK3576-C开发板 原厂的 内核/driver 进行比对，很容易知道 飞凌对 Rockchip的原厂Android14的SDK进行过魔改了！
修改之后，可以在命令行让 tinyplay播放器通过耳机发生了。
但是在Android界面还是不能播音。还需要修正！
Y:\rk3576-android14.0-20250217\kernel-6.1\sound\soc\codecs\nau8822.h
Y:\OK3576-android-source\kernel-6.1\sound\soc\codecs\nau8822.h
/* SPDX-License-Identifier: GPL-2.0 */
/*
* nau8822.h -- NAU8822 ALSA SoC Audio driver
*
* Copyright 2017 Nuvoton Technology Crop.
*
* Author: David Lin <ctlin0@nuvoton.com>
* Co-author: John Hsu <kchsu0@nuvoton.com>
* Co-author: Seven Li <wtli@nuvoton.com>
*/
#ifndef __NAU8822_H__
#define __NAU8822_H__
#include <linux/clk.h>
/* Codec Private Data */
struct nau8822 {
struct device *dev;
struct regmap *regmap;
int mclk_idx;
struct nau8822_pll pll;
int sysclk;
int div_id;
struct clk *clk;
};
Y:\rk3576-android14.0-20250217\kernel-6.1\sound\soc\codecs\nau8822.c
Y:\OK3576-android-source\kernel-6.1\sound\soc\codecs\nau8822.c
static int nau8822_i2c_probe(struct i2c_client *i2c)
{
struct device *dev = &i2c->dev;
struct nau8822 *nau8822 = dev_get_platdata(dev);
int ret;
if (!nau8822) {
nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL);
if (nau8822 == NULL)
return -ENOMEM;
}
i2c_set_clientdata(i2c, nau8822);
/* Setup clocks */
nau8822->clk = devm_clk_get(&i2c->dev, NULL);
if (IS_ERR(nau8822->clk)) {
dev_err(&i2c->dev, "codec clock missing or invalid\n");
ret = PTR_ERR(nau8822->clk);
}
ret = clk_prepare_enable(nau8822->clk);
if (ret) {
dev_err(&i2c->dev, "unable to prepare codec clk\n");
}
nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config);
if (IS_ERR(nau8822->regmap)) {
ret = PTR_ERR(nau8822->regmap);
dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
return ret;
}
nau8822->dev = dev;
/* Reset the codec */
ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822,
&nau8822_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
return ret;
}
return 0;
}
【这里还需要进一步修正！】
Y:\OK3576-android-source\hardware\rockchip\audio\tinyalsa_hal\codec_config\config_list.h
参考资料：
声卡
百度：RK3576 默认声卡
Error: 'kiss8.mp3' is not a riff/wave file
tinyplay 制定声卡
rk3588 nau8822
RK3588 nau8822
RK3576 NAU88C22YG
NAU88C22YG
BING：rk平台怎么调试nau8822
rk3576 nau8822
优快云：nau8822
https://www.nuvoton.com/resource-files/NAU8822DataSheetRev3.3.pdf
https://www.eeworld.com.cn/qrs/eic698427.html
基于RK3576开发板的声卡资源介绍
aplay -l
https://blog.youkuaiyun.com/weixin_68294039/article/details/124248608?spm=1001.2101.3001.6650.2&utm_medium=distribute.pc_relevant.none-task-blog-2%7Edefault%7EBlogCommendFromBaidu%7ECtr-2-124248608-blog-148433086.235%5Ev43%5Epc_blog_bottom_relevance_base8&depth_1-utm_source=distribute.pc_relevant.none-task-blog-2%7Edefault%7EBlogCommendFromBaidu%7ECtr-2-124248608-blog-148433086.235%5Ev43%5Epc_blog_bottom_relevance_base8&utm_relevant_index=5
RK3566,ES7202声卡驱动添加
https://blog.youkuaiyun.com/weixin_68294039/article/details/124248608
4.修改安卓hal层
在hardware/rockchip/audio/tinyalsa_hal/下的audio_hw.c,修改内容：
https://blog.youkuaiyun.com/lingzhou0909/article/details/148433086
RK3576 Android14 DMIC调制
使用tinycap工具来录音，如下
tinycap /sdcard/rec.wav -D 1 -d 0 -c 2 -r 44100 -b 16 -p 1024 -n 3
https://blog.youkuaiyun.com/qq_38312843/article/details/124868449
RK声卡调试
cat /proc/asound/cards
ls -l /dev/snd/
tinyplay /sdcard/test44.wav -D 0 -d 0 -p 1024 -n 3
hardware/rockchip/audio/tinyalsa_hal/codec_config/rt5640_config.h
【付费】
https://blog.youkuaiyun.com/radianceblau/article/details/64125411
linux驱动由浅入深系列：tinyalsa(tinymix/tinycap/tinyplay/tinypcminfo)音频子系统之一
https://blog.youkuaiyun.com/lf12345678910/article/details/102497454
tinyplay 声卡播放
https://blog.youkuaiyun.com/wenjin359/article/details/83006820
tinyalsa(tinymix/tinycap/tinyplay/tinypcminfo)
/dev/snd/pcmC*
vendor/brand/kernel/linaro/sound/core/init.c
snd_card_create
vendor/brand/kernel/linaro/sound/usb/card.c
snd_usb_audio_probe
root@:/ # cat proc/asound/cards
0 [No00 ]: brand ALSA Drv - brand MAD No.00
brand MPEG Audio Decoder No.00
root@:/ # tinypcminfo -D /proc/asound/cards
Info for card 0, device 0:
PCM out:
root@:/ # tinyplay mnt/usb/17EC-1152/file.wav -D 0 -d 2 -n 2
//tinyplay mnt/usb/17EC-1152/file.wav -D 0 -d 3 -n 2
root@:/ # ls -al dev/snd/pcm*
crw-rw---- system audio 116, 24 2007-01-01 20:00 pcmC0D0c
crw-rw---- system audio 116, 16 2007-01-01 20:00 pcmC0D0p
crw-rw---- system audio 116, 25 2007-01-01 20:00 pcmC0D1c
crw-rw---- system audio 116, 17 2007-01-01 20:00 pcmC0D1p
crw-rw---- system audio 116, 18 2007-01-01 20:00 pcmC0D2p
crw-rw---- system audio 116, 19 2007-01-01 20:00 pcmC0D3p
【付费】
https://wenku.youkuaiyun.com/answer/407dbnsft2
rk平台怎么调试nau8822 - 优快云文库.mhtml
https://blog.youkuaiyun.com/weixin_43245753/article/details/123915015
Rockchip RK3588 kernel dts解析之音频模块
https://bbs.16rd.com/thread-584429-1-1.html
[瑞芯微RK资料] 瑞芯微RK3328/RK3366/RK3368/RK3399平台Audio开发指南
2.2.4 声卡调试
1. 通过如下命令确认声卡是否注册成功
root@rk3366:/ # cat /proc/asound/cards
0 [rockchiprt5640c]: rockchip_rt5640 - rockchip,rt5640-codec
rockchip,rt5640-codec
root@rk3366:/ # ls -l /dev/snd/
crw-rw---- system audio 116, 2 2013-01-18 08:51 controlC0
crw-rw---- system audio 116, 4 2013-01-18 08:51 pcmC0D0c
crw-rw---- system audio 116, 3 2013-01-18 08:51 pcmC0D0p
2. 通过命令行播放录制调试声卡：
播放：一般播放 1khz 0db 正弦波，然后在 codec 输出端示波器简单测量是否失真，杂音，然后再使用音频分析仪测试指标。
root@rk3366:/ # tinyplay
Usage: tinyplay file.wav [-D card] [-d device] [-p period_size] [-n n_periods]
|root@rk3366:/ # tinyplay /sdcard/test44.wav -D 0 -d 0 -p 1024 -n 3
Playing sample: 2 ch, 44100 hz, 32 bit
......
https://blog.youkuaiyun.com/2301_76587520/article/details/145768380?utm_medium=distribute.pc_relevant.none-task-blog-2~default~baidujs_utm_term~default-8-145768380-blog-106573423.235^v43^pc_blog_bottom_relevance_base8&spm=1001.2101.3001.4242.5&utm_relevant_index=11
开发板适配之SAI i.MX6ULL NAU88C22【驱动driver + DTS】
SAI音频接口只是传输音频数据的数字接口，所以这些音频编解码的工作需要一个专有的编解码芯片来完成，ELF 1使用的是NAU88C22芯片，NAU88C22可连接mic和扬声器或耳机用于录音和播放声音。i.MX6ULL使用I2S总线与NAU88C22进行音频数据的传输，I2S(Inter—IC Sound)总线，又称集成电路内置音频总线，是Philips公司为数字音频设备之间的音频数据传输而制定的一种总线标准，该总线专门用于音频设备之间的数据传输，广泛应用于各种多媒体系统。
https://www.elecfans.com/d/6382666.html
RK3588核心板多媒体功能一览
先在EVM-RK3588底板接好MIC设备，再执行下面指令；采集声音 10 秒，并且保存为 wav 格式。
$ arecord -D default:CARD=rockchipnau8822 -d 10 -f cd -t wav test1.wav
https://bbs.elecfans.com/jishu_2477148_1_1.html
[活动] 【ELF 2学习板试用】05 ASoC中的machine class
扫一扫
697
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
