hal: Reduce debug logs
Remove or mute debug logs that are not necessary in debugging
usual audio HAL issues.
Change-Id: I832d9d95b6b24f64871524efbe65dd57850afd41
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index d53db94..783d37f 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -4038,7 +4038,7 @@
void audio_extn_fm_get_parameters(struct str_parms *query, struct str_parms *reply)
{
if(audio_extn_fm_power_opt_enabled) {
- ALOGD("%s: Enter", __func__);
+ ALOGV("%s: Enter", __func__);
fm_get_parameters(query, reply);
}
}
@@ -4047,7 +4047,7 @@
struct str_parms *parms)
{
if(audio_extn_fm_power_opt_enabled) {
- ALOGD("%s: Enter", __func__);
+ ALOGV("%s: Enter", __func__);
fm_set_parameters(adev, parms);
}
}
@@ -5626,7 +5626,7 @@
return;
} else {
mixer_ctl_set_value(ctl, 0, value);
- ALOGD("%s: mixer_value set %d", __func__, value);
+ ALOGV("%s: mixer_value set %d", __func__, value);
}
return;
}
diff --git a/hal/audio_extn/device_utils.c b/hal/audio_extn/device_utils.c
index a0ee5b5..e1736ef 100644
--- a/hal/audio_extn/device_utils.c
+++ b/hal/audio_extn/device_utils.c
@@ -503,7 +503,6 @@
goto done;
if (type == AUDIO_DEVICE_NONE) {
- ALOGE("%s: Invalid device: %#x", __func__, type);
ret = -EINVAL;
goto done;
}
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 3b084ae..bbdbfb5 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -545,7 +545,7 @@
float vol;
char value[32]={0};
- ALOGD("%s: enter", __func__);
+ ALOGV("%s: enter", __func__);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_HFP_ENABLE, value,
sizeof(value));
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 04f2268..52dd070 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -600,10 +600,8 @@
uc_info->in_snd_device < SND_DEVICE_IN_END)) {
if (is_same_as_st_device(uc_info->in_snd_device))
update_device_list(&ev_info.device_info.devices, ST_DEVICE_HANDSET_MIC, "", true);
- } else {
- ALOGE("%s: invalid input device 0x%x, for event %d",
- __func__, uc_info->in_snd_device, event);
}
+
raise_event = platform_sound_trigger_usecase_needs_event(uc_info->id);
ALOGD("%s: uc_info->id %d of type %d for Event %d, with Raise=%d",
__func__, uc_info->id, uc_info->type, event, raise_event);
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 8af4d80..6af81ca 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -2179,10 +2179,8 @@
int err = 0;
char value[MAX_STR_SIZE] = {0};
- if (!handle.spkr_prot_enable) {
- ALOGD("%s: Speaker protection disabled", __func__);
+ if (!handle.spkr_prot_enable)
return -EINVAL;
- }
err = str_parms_get_str(query, AUDIO_PARAMETER_KEY_FBSP_GET_SPKR_CAL, value,
sizeof(value));
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 5f10ca7..d396ae3 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -767,7 +767,8 @@
else if (-ENOSYS == bw)
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
- ALOGI("%s Allowing 24 and above bits playback on speaker ONLY at default sampling rate", __func__);
+ ALOGV("%s Allowing 24 and above bits playback on speaker \
+ ONLY at default sampling rate", __func__);
}
property_get("vendor.audio.playback.mch.downsample",value,"");
@@ -887,7 +888,7 @@
*sample_rate = 16000;
break;
default:
- ALOGD("%s:Not a BT SCO device, need not update sampling rate\n", __func__);
+ ALOGV("%s:Not a BT SCO device, need not update sampling rate\n", __func__);
break;
}
}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 0594b7b..5ee30c4 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -728,7 +728,7 @@
if (stream_info != NULL)
free(stream_info);
pthread_mutex_unlock(&adev->lock);
- ALOGD("%s: Added stream in io_streams_map with handle %d", __func__, handle);
+ ALOGV("%s: Added stream in io_streams_map with handle %d", __func__, handle);
return 0;
}
@@ -740,7 +740,7 @@
hashmapRemove(adev->io_streams_map, (void *) (intptr_t) handle);
if (s_info == NULL)
goto done;
- ALOGD("%s: Removed stream with handle %d", __func__, handle);
+ ALOGV("%s: Removed stream with handle %d", __func__, handle);
patch_map_remove_l(adev, s_info->patch_handle);
free(s_info);
done:
@@ -994,15 +994,15 @@
if((new_backend_idx == HEADPHONE_BACKEND) &&
((usecase_backend_idx == HEADPHONE_44_1_BACKEND) ||
(usecase_backend_idx == DSD_NATIVE_BACKEND))) {
- ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
+ ALOGV("%s:DSD or native stream detected enabling asrcmode in hardware",
__func__);
enable_asrc_mode(adev);
break;
} else if(((new_backend_idx == HEADPHONE_44_1_BACKEND) ||
(new_backend_idx == DSD_NATIVE_BACKEND)) &&
(usecase_backend_idx == HEADPHONE_BACKEND)) {
- ALOGD("%s:48K stream detected, disabling and enabling it with asrcmode in hardware",
- __func__);
+ ALOGV("%s: 48K stream detected, disabling and enabling it \
+ with asrcmode in hardware", __func__);
disable_audio_route(adev, uc);
disable_snd_device(adev, uc->out_snd_device);
// Apply true-high-quality-mode if DSD or > 44.1KHz or >=24-bit
@@ -3067,7 +3067,7 @@
&voip_out->app_type_cfg.gain[0]);
}
- ALOGD("%s: done",__func__);
+ ALOGV("%s: done",__func__);
return status;
}
@@ -3312,7 +3312,7 @@
done_open:
audio_streaming_hint_end();
audio_extn_perf_lock_release(&adev->perf_lock_handle);
- ALOGD("%s: exit", __func__);
+ ALOGV("%s: exit", __func__);
enable_gcov();
return ret;
@@ -4151,7 +4151,7 @@
}
audio_streaming_hint_end();
audio_extn_perf_lock_release(&adev->perf_lock_handle);
- ALOGD("%s: exit", __func__);
+ ALOGV("%s: exit", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_ULL ||
out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
@@ -4582,7 +4582,7 @@
pthread_mutex_unlock(&adev->lock);
}
pthread_mutex_unlock(&out->lock);
- ALOGD("%s: exit", __func__);
+ ALOGV("%s: exit", __func__);
return 0;
}
@@ -4676,7 +4676,7 @@
stop_output_stream(out);
ATRACE_END();
}
- ALOGD("%s: exit", __func__);
+ ALOGV("%s: exit", __func__);
return 0;
}
@@ -6926,7 +6926,7 @@
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
- ALOGD("%s: exit: status(%d)", __func__, ret);
+ ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
@@ -8992,7 +8992,7 @@
vr_audio_enabled = adev->vr_audio_mode_enabled;
pthread_mutex_unlock(&adev->lock);
- ALOGI("getting vr mode to %d", vr_audio_enabled);
+ ALOGV("getting vr mode to %d", vr_audio_enabled);
if (vr_audio_enabled) {
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
@@ -9018,7 +9018,7 @@
str_parms_destroy(query);
str_parms_destroy(reply);
- ALOGD("%s: exit: returns - %s", __func__, str);
+ ALOGV("%s: exit: returns - %s", __func__, str);
return str;
}
@@ -9900,7 +9900,7 @@
struct audio_patch_info *p_info,
patch_type_t patch_type, bool new_patch)
{
- ALOGD("%s: enter", __func__);
+ ALOGV("%s: enter", __func__);
if (p_info == NULL) {
ALOGE("%s: Invalid patch pointer", __func__);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 0be37a4..c8d355f 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -3047,11 +3047,8 @@
for (i = 0; i < SND_DEVICE_MAX; i++) {
valid_hw_interface = false;
- if (hw_interface_table[i] == NULL) {
- ALOGW("%s: sound device %s has no hw interface set\n",
- __func__, platform_get_snd_device_name(i));
+ if (hw_interface_table[i] == NULL)
continue;
- }
for (j = 0; j < max_be_dai_names; j++) {
if (strcmp(hw_interface_table[i], be_dai_name_table[j].be_name)
@@ -3060,9 +3057,6 @@
break;
}
}
- if (!valid_hw_interface)
- ALOGD("%s: sound device %s does not have a valid hw interface set (disregard for combo devices) %s\n",
- __func__, platform_get_snd_device_name(i), hw_interface_table[i]);
}
goto done;
@@ -3490,13 +3484,10 @@
#endif
/* CSRA devices support multiple sample rates via I2S at spkr out */
- if (!strncmp(snd_card_name, "qcs405-csra", strlen("qcs405-csra"))) {
- ALOGE("%s: soundcard: %s supports multiple sample rates", __func__, snd_card_name);
+ if (!strncmp(snd_card_name, "qcs405-csra", strlen("qcs405-csra")))
my_data->use_sprk_default_sample_rate = false;
- } else {
+ else
my_data->use_sprk_default_sample_rate = true;
- ALOGE("%s: soundcard: %s supports only default sample rate", __func__, snd_card_name);
- }
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
@@ -4039,7 +4030,7 @@
}
}
}
- ALOGI("%s: no matching param with id %d ip_ch %d op_ch %d uc_id %d snd_dev %d",
+ ALOGV("%s: no matching param with id %d ip_ch %d op_ch %d uc_id %d snd_dev %d",
__func__, info->id, info->ip_channels, info->op_channels,
info->usecase_id[0], info->snd_device);
return NULL;
@@ -4431,12 +4422,12 @@
platform_get_snd_device_name(snd_device2));
if ((snd_device1 < SND_DEVICE_MIN) || (snd_device1 >= SND_DEVICE_OUT_END)) {
- ALOGE("%s: Invalid snd_device = %s", __func__,
+ ALOGV("%s: Invalid snd_device1 = %s", __func__,
platform_get_snd_device_name(snd_device1));
return false;
}
if ((snd_device2 < SND_DEVICE_MIN) || (snd_device2 >= SND_DEVICE_OUT_END)) {
- ALOGE("%s: Invalid snd_device = %s", __func__,
+ ALOGV("%s: Invalid snd_device2 = %s", __func__,
platform_get_snd_device_name(snd_device2));
return false;
}
@@ -6118,7 +6109,7 @@
}
- ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
+ ALOGV("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
snd_device, *num_devices, *new_snd_devices);
return ret;
@@ -8057,7 +8048,7 @@
}
}
cal.acdb_dev_id = platform_get_snd_device_acdb_id(cal.snd_dev_id);
- ALOGD("Setting audio calibration for snd_device(%d) acdb_id(%d)",
+ ALOGV("Setting audio calibration for snd_device(%d) acdb_id(%d)",
cal.snd_dev_id, cal.acdb_dev_id);
if(cal.acdb_dev_id == -EINVAL) {
ALOGE("[%s] Invalid acdb_device id %d for snd device id %d",
@@ -8812,7 +8803,8 @@
for (i = 0; i < sizeof(dsp_only_decoders_mime)/sizeof(dsp_only_decoders_mime[0]); i++) {
if (!strncmp(decoder_mime_type, dsp_only_decoders_mime[i],
strlen(dsp_only_decoders_mime[i]))) {
- ALOGD("Rejecting request for DSP only session from HAL during voice call/SSR state");
+ ALOGV("Rejecting request for DSP only session from HAL \
+ during voice call/SSR state");
isallowed = 0;
break;
}
@@ -10097,7 +10089,7 @@
*/
if (platform_spkr_use_default_sample_rate(adev->platform)) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
+ ALOGV("%s:becf: afe: playback on codec device not supporting native playback set "
"default Sample Rate(48k)", __func__);
}
}
diff --git a/hal/voice.c b/hal/voice.c
index 72c3372..034eaa0 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -435,7 +435,7 @@
bool in_call_rec = false;
if (!in) {
- ALOGE("%s: input stream is NULL", __func__);
+ ALOGV("%s: input stream is NULL", __func__);
return in_call_rec;
}