Merge "hal: Avoid sending MAD calibration"
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 5ea5a43..e703b13 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -485,6 +485,7 @@
enable_audio_route(adev, uc_info_tx);
pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
+ ALOGV("%s: pcm device id %d", __func__, pcm_dev_tx_id);
if (pcm_dev_tx_id < 0) {
ALOGE("%s: Invalid pcm device for usecase (%d)",
__func__, uc_info_tx->id);
@@ -647,6 +648,7 @@
char wsa_path[MAX_PATH] = {0};
int spk_1_tzn, spk_2_tzn;
char buf[32] = {0};
+ int ret;
/* If the value of this persist.spkr.cal.duration is 0
* then it means it will take 30min to calibrate
@@ -756,13 +758,17 @@
thermal_fd = open(wsa_path, O_RDONLY);
if (thermal_fd > 0) {
for (i = 0; i < NUM_ATTEMPTS; i++) {
- if (read(thermal_fd, buf, sizeof(buf))) {
+ if ((ret = read(thermal_fd, buf, sizeof(buf))) >= 0) {
t0_spk_1 = atoi(buf);
- if (i > 0 && (t0_spk_1 != t0_spk_prior))
+ if (i > 0 && (t0_spk_1 != t0_spk_prior)) {
+ ALOGE("%s: spkr1 curr temp: %d, prev temp: %d\n",
+ __func__, t0_spk_1, t0_spk_prior);
break;
+ }
t0_spk_prior = t0_spk_1;
+ sleep(1);
} else {
- ALOGE("%s: read fail for %s\n", __func__, wsa_path);
+ ALOGE("%s: read fail for %s err:%d\n", __func__, wsa_path, ret);
break;
}
}
@@ -773,26 +779,30 @@
if (i == NUM_ATTEMPTS) {
/*Convert temp into q6 format*/
t0_spk_1 = (t0_spk_1 * (1 << 6));
- ALOGE("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
+ ALOGD("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
} else {
- ALOGE("%s: thermal equilibrium failed for spkr1 in %d readings\n",
- __func__, NUM_ATTEMPTS);
+ ALOGD("%s: thermal equilibrium failed for spkr1 in %d/%d readings\n",
+ __func__, i, NUM_ATTEMPTS);
t0_spk_1 = SAFE_SPKR_TEMP_Q6;
}
}
if (spk_2_tzn >= 0) {
snprintf(wsa_path, MAX_PATH, TZ_WSA, spk_2_tzn);
- ALOGE("%s: wsa_path: %s\n", __func__, wsa_path);
+ ALOGD("%s: wsa_path: %s\n", __func__, wsa_path);
thermal_fd = open(wsa_path, O_RDONLY);
if (thermal_fd > 0) {
for (i = 0; i < NUM_ATTEMPTS; i++) {
- if (read(thermal_fd, buf, sizeof(buf))) {
+ if ((ret = read(thermal_fd, buf, sizeof(buf))) >= 0) {
t0_spk_2 = atoi(buf);
- if (i > 0 && (t0_spk_2 != t0_spk_prior))
+ if (i > 0 && (t0_spk_2 != t0_spk_prior)) {
+ ALOGE("%s: spkr2 curr temp: %d, prev temp: %d\n",
+ __func__, t0_spk_2, t0_spk_prior);
break;
+ }
t0_spk_prior = t0_spk_2;
+ sleep(1);
} else {
- ALOGE("%s: read fail for %s\n", __func__, wsa_path);
+ ALOGE("%s: read fail for %s err:%d\n", __func__, wsa_path, ret);
break;
}
}
@@ -803,10 +813,10 @@
if (i == NUM_ATTEMPTS) {
/*Convert temp into q6 format*/
t0_spk_2 = (t0_spk_2 * (1 << 6));
- ALOGE("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
+ ALOGD("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
} else {
- ALOGE("%s: thermal equilibrium failed for spkr2 in %d readings\n",
- __func__, NUM_ATTEMPTS);
+ ALOGE("%s: thermal equilibrium failed for spkr2 in %d/%d readings\n",
+ __func__, i, NUM_ATTEMPTS);
t0_spk_2 = SAFE_SPKR_TEMP_Q6;
}
}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 80f56a4..c67f210 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -252,6 +252,26 @@
static int set_voice_volume_l(struct audio_device *adev, float volume);
+__attribute__ ((visibility ("default")))
+bool audio_hw_send_gain_dep_calibration(int level) {
+ bool ret_val = false;
+ ALOGV("%s: called ... ", __func__);
+
+ pthread_mutex_lock(&adev_init_lock);
+
+ if (adev != NULL && adev->platform != NULL) {
+ pthread_mutex_lock(&adev->lock);
+ ret_val = platform_send_gain_dep_cal(adev->platform, level);
+ pthread_mutex_unlock(&adev->lock);
+ } else {
+ ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform");
+ }
+
+ pthread_mutex_unlock(&adev_init_lock);
+
+ return ret_val;
+}
+
static int check_and_set_gapless_mode(struct audio_device *adev) {
@@ -2182,7 +2202,7 @@
if (ret != 0) {
if (out->pcm)
- ALOGE("%s: error %ld - %s", __func__, ret, pcm_get_error(out->pcm));
+ ALOGE("%s: error %zu - %s", __func__, ret, pcm_get_error(out->pcm));
if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
voice_extn_compress_voip_close_output_stream(&out->stream.common);
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 3055772..9d019c5 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -3289,6 +3289,11 @@
return fragment_size;
}
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+ return OFFLOAD_USE_SMALL_BUFFER;
+}
+
int platform_set_codec_backend_cfg(struct audio_device* adev,
unsigned int bit_width, unsigned int sample_rate)
{
@@ -4127,3 +4132,9 @@
done:
return ret;
}
+
+bool platform_send_gain_dep_cal(void *platform __unused,
+ int level __unused)
+{
+ return 0;
+}
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 1d5d1e6..861790e 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1171,3 +1171,9 @@
{
return -ENOSYS;
}
+
+bool platform_send_gain_dep_cal(void *platform __unused,
+ int level __unused)
+{
+ return 0;
+}
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 7ba93b0..3c90a8e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -54,6 +54,7 @@
#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/system/etc/audio_platform_info_i2s.xml"
+#include <linux/msm_audio.h>
#define LIB_ACDB_LOADER "libacdbloader.so"
#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
@@ -155,6 +156,12 @@
uint32_t param_id;
} acdb_audio_cal_cfg_t;
+enum {
+ CAL_MODE_SEND = 0x1,
+ CAL_MODE_PERSIST = 0x2,
+ CAL_MODE_RTAC = 0x4
+};
+
/* Audio calibration related functions */
typedef void (*acdb_deallocate_t)();
typedef int (*acdb_init_t)(const char *, char *, int);
@@ -179,6 +186,7 @@
bool ui_na_prop_enabled;
} native_audio_prop;
static native_audio_prop na_props = {0, 0};
+typedef int (*acdb_send_gain_dep_cal_t)(int, int, int, int, int);
struct platform_data {
struct audio_device *adev;
@@ -209,6 +217,7 @@
acdb_get_default_app_type_t acdb_get_default_app_type;
void *hw_info;
+ acdb_send_gain_dep_cal_t acdb_send_gain_dep_cal;
struct csd_data *csd;
void *edid_info;
bool edid_valid;
@@ -704,6 +713,55 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+bool platform_send_gain_dep_cal(void *platform, int level) {
+ bool ret_val = false;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct audio_device *adev = my_data->adev;
+ int acdb_dev_id, app_type;
+ int acdb_dev_type = MSM_SNDDEV_CAP_RX;
+ int mode = CAL_MODE_RTAC;
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ if (my_data->acdb_send_gain_dep_cal == NULL) {
+ ALOGE("%s: dlsym error for acdb_send_gain_dep_cal", __func__);
+ return ret_val;
+ }
+
+ if (!voice_is_in_call(adev)) {
+ ALOGV("%s: Not Voice call usecase, apply new cal for level %d",
+ __func__, level);
+
+ // find the current active sound device
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+
+ if (usecase != NULL &&
+ usecase->type == PCM_PLAYBACK &&
+ (usecase->stream.out->devices == AUDIO_DEVICE_OUT_SPEAKER ||
+ usecase->stream.out->devices == AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
+
+ ALOGV("%s: out device is %d", __func__, usecase->out_snd_device);
+ app_type = usecase->stream.out->app_type_cfg.app_type;
+ acdb_dev_id = acdb_device_table[usecase->out_snd_device];
+ if (!my_data->acdb_send_gain_dep_cal(acdb_dev_id, app_type,
+ acdb_dev_type, mode, level)) {
+ // set ret_val true if at least one calibration is set successfully
+ ret_val = true;
+ } else {
+ ALOGE("%s: my_data->acdb_send_gain_dep_cal failed ", __func__);
+ }
+ } else {
+ ALOGW("%s: Usecase list is empty", __func__);
+ }
+ }
+ } else {
+ ALOGW("%s: Voice call in progress .. ignore setting new cal",
+ __func__);
+ }
+ return ret_val;
+}
+
void platform_set_echo_reference(void *platform, bool enable)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -1254,6 +1312,12 @@
ALOGE("%s: Could not find the symbol acdb_get_default_app_type from %s",
__func__, LIB_ACDB_LOADER);
+ my_data->acdb_send_gain_dep_cal = (acdb_send_gain_dep_cal_t)dlsym(my_data->acdb_handle,
+ "acdb_loader_send_gain_dep_cal");
+ if (!my_data->acdb_send_gain_dep_cal)
+ ALOGV("%s: Could not find the symbol acdb_loader_send_gain_dep_cal from %s",
+ __func__, LIB_ACDB_LOADER);
+
my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
"acdb_loader_init_v2");
if (my_data->acdb_init == NULL) {
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 2618bd8..2247933 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -227,6 +227,8 @@
#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
#ifdef PLATFORM_APQ8084
#define SPKR_PROT_CALIB_TX_PCM_DEVICE 35
+#elif PLATFORM_MSM8996
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 42
#else
#define SPKR_PROT_CALIB_TX_PCM_DEVICE 25
#endif
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 59cb554..f8749e7 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -30,6 +30,7 @@
int platform_get_snd_device_name_extn(void *platform, snd_device_t snd_device,
char *device_name);
void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
+bool platform_send_gain_dep_cal(void *platform, int level);
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
int platform_get_snd_device_index(char *snd_device_index_name);
int platform_set_fluence_type(void *platform, char *value);