功率调试,是1Khz 0db的源将soc dsp到pa的幅度降到一样(理想状态,调到dsp最大幅度),再调pa相对增益以达到特定功率(这里为5W),只需要调试这个功率的相对增益就行,设置到pa之后再调dsp的幅度,每个幅度都会做相应的增益补偿(与最大功率一样)
整个调试过程中比较纠结,到最后才发现i2s bit clock mclock波形质量不好
且模块需要在提供i2s clock的时候,读写才有效
在调试音频功率时发现,做line in时,pm8916有异常波形出来,需要削除掉这个异常15k正弦波形
后面是发现是从mic1进来的,由硬件进行滤波来解决这个问题,后面在解决中还出现了23k正弦波,同样是硬件解决的
最后硬件只将tinymix “ADC1 Volume” 5就可以调pa的功率为5w
只需在mix_path将默认值修改为5,再将ftm_test_config配置文件中去掉ADC1 Volume:6的设置
Index: android/hardware/qcom/audio/configs/msm8909/mixer_paths_msm8909_pm8916.xml
===================================================================
--- android/hardware/qcom/audio/configs/msm8909/mixer_paths_msm8909_pm8916.xml (revision 281)
+++ android/hardware/qcom/audio/configs/msm8909/mixer_paths_msm8909_pm8916.xml (revision 282)
@@ -56,7 +56,7 @@
<ctl name="IIR1 INP2 Volume" value="84" />
<ctl name="IIR1 INP3 Volume" value="84" />
<ctl name="IIR1 INP4 Volume" value="84" />
- <ctl name="ADC1 Volume" value="4" />
+ <ctl name="ADC1 Volume" value="5" />
<ctl name="ADC2 Volume" value="4" />
<ctl name="ADC3 Volume" value="4" />
<ctl name="DEC1 Volume" value="84" />
Index: android/vendor/qcom/proprietary/mm-audio/audio_ftm/family-b/config/8909/ftm_test_config
===================================================================
--- android/vendor/qcom/proprietary/mm-audio/audio_ftm/family-b/config/8909/ftm_test_config (revision 281)
+++ android/vendor/qcom/proprietary/mm-audio/audio_ftm/family-b/config/8909/ftm_test_config (revision 282)
@@ -802,8 +802,6 @@
DEC1 MUX:ADC1
-ADC1 Volume:6
-
QUAT_MI2S_RX Port Mixer TERT_MI2S_TX:1
QUAT_MI2S_RX_DL_HL Switch:1
在调试过程中,还发现
针对tas5782m模块
ppc3仿真工具操作不方便，数据误发
写的i2c地址是0x92其实是0x90需要手动再发一现，非常麻烦，
导出来的寄存器数据有误
导出来的数据发现有问题，有一定的风险
如下导出来是
{ 0x22, 0x03 },实现操作的是96k
后来发现ppc3无法加载单声道,此时pa是采用声道设计的,且原来i2c地址采用的0x90
原来关于tas5782m,TI SW(ppc3)内部已经固定的立体声设备地址为0x90而Mono设备为0x92，这就是无法选择相应的单声道模式的原因
从上说0x90是立体声道,因此需要硬件将i2c 地址修改为0x92,软件也需要修改
因为默认为低,因此adr0必须拉高,且驱动将i2c地址修改为0x49
重新修改后可以识别了,重新用ppc3调试
在没有连接的状态下,在recent files选 一个ppc3文件
点end system integration,确保一下参数是否与调试相匹配
点左下角的连接,
点系统检测
硬件检测后喇叭就会发出声音,检测完成后就没有声音,说明硬件检测通了,可以开始调试了
回到home进入audio processing音频调试界面
重新调96K
导出寄存器表
这里导出表的头文件内容类型为,如果ppc3软件有问题,还需要根据spec修改寄存器里面的信息,如clk源,i2s 格式,电压等
调节音量直接走调pa的相对音量增益
在高通的audio hal 中修改即文件 audio_hw.c的out_set_volume中
#if 1
#define ESP (0.000001)
static float volume_value[]=
{
0.000000,
0.002701,
0.006464,
0.013924,
0.023646,
0.040156,
0.072862,
0.123738,
0.189453,
0.294274,
0.435262,
0.643799,
1.000000,
};//打印出来用于定位到声音处于哪一阶
#endif
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
struct stream_out *out = (struct stream_out *)stream;
int volume[2];
int ret = 0;
ALOGE("[Linda]11 %s: left:%f, right: %f, out->usecase:%d",__func__, left, right,out->usecase);
if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
/* only take left channel into account: the API is for stereo anyway */
out->muted = (left == 0.0f);
return 0;
} else if (is_offload_usecase(out->usecase)) {
if (audio_extn_passthru_is_passthrough_stream(out)) {
/*
* Set mute or umute on HDMI passthrough stream.
* Only take left channel into account.
* Mute is 0 and unmute 1
*/
audio_extn_passthru_set_volume(out, (left == 0.0f));
} else if (out->format == AUDIO_FORMAT_DSD){
char mixer_ctl_name[128] = "DSD Volume";
struct audio_device *adev = out->dev;
struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
volume[0] = (int)(AmpToDb(left));
volume[1] = (int)(AmpToDb(right));
mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
return 0;
} else {
ALOGE("[Linda]77 %s:",__func__);
pthread_mutex_lock(&out->compr_mute_lock);
ALOGE("%s: compress mute %d", __func__, out->a2dp_compress_mute);
#if 0
if (!out->a2dp_compress_mute)
ret = out_set_compr_volume(stream, left, right);//注释掉这里正常播放的音量全部都是在这里设到adsp
#else
{
#if 0
FILE * fd;
unsigned int value =10;
ALOGE("%s: start test", __func__);
fd = fopen("/sys/devices/soc/78b8000.i2c/i2c-4/4-0048/pa_vol","rw");
if (fd == NULL)
{
ALOGE("%s: start test fd == null", __func__);
//return -1;
}
ALOGE("%s: start wrtie", __func__);
if(fd != NULL)
{
ALOGE("%s: start wrtie 0", __func__);
fwrite(&value, sizeof(value), 1, fd);
ALOGE("%s: start wrtie 1", __func__);
}
fclose(fd);
#else
ALOGE("%s: start test", __func__);
int i =0;
char devicepath[512] = {0};
for(i = 0; i < 12; i++)
{
ALOGE("%s: volume_value[i]: %f,right:%f,left:%f", __func__,volume_value[i],right,left);
if(((volume_value[i] - left) > (-1) * ESP) && ((volume_value[i] - left) < ESP) &&
((volume_value[i] - right) > (-1) * ESP) && ((volume_value[i] - right) < ESP) )
{
sprintf(devicepath, "echo %d > /sys/devices/soc/78b8000.i2c/i2c-4/4-0048/pa_vol", i);
ALOGE("%s: test command: %s", __func__,devicepath);
system(devicepath);//将音量除数写到pa中，驱动层直接控制音量的相对增益寄存器
break;
}
}
#endif
ALOGE("%s: test end", __func__);
}
#endif
out->volume_l = left;
out->volume_r = right;
pthread_mutex_unlock(&out->compr_mute_lock);
return ret;
}
} else if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP) {
char mixer_ctl_name[] = "App Type Gain";
struct audio_device *adev = out->dev;
struct mixer_ctl *ctl;
uint32_t set_values[4];
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
set_values[0] = 0; //0: Rx Session 1:Tx Session
set_values[1] = out->app_type_cfg.app_type;
set_values[2] = (int)(left * VOIP_PLAYBACK_VOLUME_MAX);
set_values[3] = (int)(right * VOIP_PLAYBACK_VOLUME_MAX);
mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
return 0;
}
return -ENOSYS;
}
但后来发现以上的修改只能对播放mp3之类的有效果，对wav类的pcm数据没有影响，后来发现是audioflinger针对不同的流在混音时做了不同的处理，需要在AudioFlinger::setStreamVolume中作处理
https://blog.csdn.net/yangchangwen/article/details/78348317
