Merge "post_proc: fix noise at start of multichannel playback in hw_acc mode"
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 8601a7e..f51cfa7 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -58,6 +58,10 @@
#define SAFE_SPKR_TEMP 40
#define SAFE_SPKR_TEMP_Q6 (SAFE_SPKR_TEMP * (1 << 6))
+/*Bongo Spkr temp range*/
+#define TZ_TEMP_MIN_THRESHOLD (5)
+#define TZ_TEMP_MAX_THRESHOLD (45)
+
/*Range of resistance values 2ohms to 40 ohms*/
#define MIN_RESISTANCE_SPKR_Q24 (2 * (1 << 24))
#define MAX_RESISTANCE_SPKR_Q24 (40 * (1 << 24))
@@ -214,7 +218,8 @@
snprintf(name, MAX_PATH, TZ_TYPE, tzn);
ALOGD("Opening %s\n", name);
read_line_from_file(name, buf, sizeof(buf));
- buf[strlen(buf)] = '\0';
+ if (strlen(buf) > 0)
+ buf[strlen(buf) - 1] = '\0';
if (!strcmp(buf, sensor_name)) {
found = 1;
break;
@@ -726,24 +731,23 @@
close(acdb_fd);
}
+ ALOGV("%s: start calibration", __func__);
while (1) {
- ALOGV("%s: start calibration", __func__);
if (handle.wsa_found) {
spk_1_tzn = handle.spkr_1_tzn;
spk_2_tzn = handle.spkr_2_tzn;
goahead = false;
pthread_mutex_lock(&adev->lock);
if (is_speaker_in_use(&sec)) {
- ALOGD("%s: WSA Speaker in use retry calibration", __func__);
+ ALOGV("%s: WSA Speaker in use retry calibration", __func__);
pthread_mutex_unlock(&adev->lock);
continue;
} else {
- ALOGD("%s: wsa speaker idle %ld min time %ld", __func__, sec, min_idle_time);
if (sec < min_idle_time) {
- ALOGD("%s: speaker idle is less retry", __func__);
pthread_mutex_unlock(&adev->lock);
continue;
}
+ ALOGV("%s: wsa speaker idle %ld min time %ld", __func__, sec, min_idle_time);
goahead = true;
}
if (!list_empty(&adev->usecase_list)) {
@@ -753,7 +757,7 @@
if (goahead) {
if (spk_1_tzn >= 0) {
snprintf(wsa_path, MAX_PATH, TZ_WSA, spk_1_tzn);
- ALOGD("%s: wsa_path: %s\n", __func__, wsa_path);
+ ALOGV("%s: wsa_path: %s\n", __func__, wsa_path);
thermal_fd = -1;
thermal_fd = open(wsa_path, O_RDONLY);
if (thermal_fd > 0) {
@@ -777,18 +781,24 @@
ALOGE("%s: fd for %s is NULL\n", __func__, wsa_path);
}
if (i == NUM_ATTEMPTS) {
+ if (t0_spk_1 < TZ_TEMP_MIN_THRESHOLD ||
+ t0_spk_1 > TZ_TEMP_MAX_THRESHOLD) {
+ pthread_mutex_unlock(&adev->lock);
+ continue;
+ }
/*Convert temp into q6 format*/
t0_spk_1 = (t0_spk_1 * (1 << 6));
ALOGD("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
} else {
- ALOGD("%s: thermal equilibrium failed for spkr1 in %d/%d readings\n",
+ ALOGV("%s: thermal equilibrium failed for spkr1 in %d/%d readings\n",
__func__, i, NUM_ATTEMPTS);
- t0_spk_1 = SAFE_SPKR_TEMP_Q6;
+ pthread_mutex_unlock(&adev->lock);
+ continue;
}
}
if (spk_2_tzn >= 0) {
snprintf(wsa_path, MAX_PATH, TZ_WSA, spk_2_tzn);
- ALOGD("%s: wsa_path: %s\n", __func__, wsa_path);
+ ALOGV("%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++) {
@@ -811,13 +821,19 @@
ALOGE("%s: fd for %s is NULL\n", __func__, wsa_path);
}
if (i == NUM_ATTEMPTS) {
+ if (t0_spk_2 < TZ_TEMP_MIN_THRESHOLD ||
+ t0_spk_2 > TZ_TEMP_MAX_THRESHOLD) {
+ pthread_mutex_unlock(&adev->lock);
+ continue;
+ }
/*Convert temp into q6 format*/
t0_spk_2 = (t0_spk_2 * (1 << 6));
ALOGD("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
} else {
- ALOGE("%s: thermal equilibrium failed for spkr2 in %d/%d readings\n",
+ ALOGV("%s: thermal equilibrium failed for spkr2 in %d/%d readings\n",
__func__, i, NUM_ATTEMPTS);
- t0_spk_2 = SAFE_SPKR_TEMP_Q6;
+ pthread_mutex_unlock(&adev->lock);
+ continue;
}
}
}
@@ -848,16 +864,15 @@
goahead = false;
pthread_mutex_lock(&adev->lock);
if (is_speaker_in_use(&sec)) {
- ALOGD("%s: Speaker in use retry calibration", __func__);
+ ALOGV("%s: Speaker in use retry calibration", __func__);
pthread_mutex_unlock(&adev->lock);
continue;
} else {
- ALOGD("%s: speaker idle %ld min time %ld", __func__, sec, min_idle_time);
if (sec < min_idle_time) {
- ALOGD("%s: speaker idle is less retry", __func__);
pthread_mutex_unlock(&adev->lock);
continue;
}
+ ALOGD("%s: speaker idle %ld min time %ld", __func__, sec, min_idle_time);
goahead = true;
}
if (!list_empty(&adev->usecase_list)) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 4bc6b59..e50e1b6 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -718,7 +718,8 @@
if (usecase->type != PCM_PLAYBACK &&
usecase != uc_info &&
usecase->in_snd_device != snd_device &&
- (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ ((uc_info->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
(usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
@@ -2802,7 +2803,8 @@
out->use_small_bufs = false;
/* Init use case and pcm_config */
- if ((out->flags == AUDIO_OUTPUT_FLAG_DIRECT) &&
+ if ((out->flags & AUDIO_OUTPUT_FLAG_DIRECT) &&
+ !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ||
out->devices & AUDIO_DEVICE_OUT_PROXY)) {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 3a8f314..02cb673 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -1237,6 +1237,10 @@
const char *snd_card_name;
char mixer_xml_path[100],ffspEnable[PROPERTY_VALUE_MAX];
char *cvd_version = NULL;
+ int idx;
+
+ my_data = calloc(1, sizeof(struct platform_data));
+
if (!my_data) {
ALOGE("failed to allocate platform data");
return NULL;
@@ -1760,7 +1764,7 @@
{
if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
ALOGE("%s: Invalid snd_device = %d", __func__, snd_device);
- return DEFAULT_OUTPUT_SAMPLING_RATE;
+ return CODEC_BACKEND_DEFAULT_BIT_WIDTH;
}
return backend_bit_width_table[snd_device];
}
@@ -3695,6 +3699,16 @@
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
}
+
+ /*
+ * Sample rate greater than 48K is only supported by external codecs on
+ * specific devices e.g. Headphones, reset the sample rate to
+ * default value if not external codec.
+ */
+ if (!is_external_codec)
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+
+
ALOGI("%s Codec selected backend: %d updated bit width: %d and sample rate: %d",
__func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index c63bfe9..af2f6ae 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -47,6 +47,15 @@
(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)
+/*
+ * Below are the input devices for which back end is same, SLIMBUS_0_TX.
+ * All these devices are handled by the internal HW codec. We can
+ * enable any one of these devices at any time
+ */
+#define AUDIO_DEVICE_IN_ALL_CODEC_BACKEND \
+ (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC | \
+ AUDIO_DEVICE_IN_WIRED_HEADSET | AUDIO_DEVICE_IN_VOICE_CALL) & ~AUDIO_DEVICE_BIT_IN
+
/* Sound devices specific to the platform
* The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
* devices to enable corresponding mixer paths
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index 4b4d14e..aab5436 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013, 2015 The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -35,6 +35,15 @@
(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)
+/*
+ * Below are the input devices for which back end is same, SLIMBUS_0_TX.
+ * All these devices are handled by the internal HW codec. We can
+ * enable any one of these devices at any time
+ */
+#define AUDIO_DEVICE_IN_ALL_CODEC_BACKEND \
+ (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC | \
+ AUDIO_DEVICE_IN_WIRED_HEADSET | AUDIO_DEVICE_IN_VOICE_CALL) & ~AUDIO_DEVICE_BIT_IN
+
/* Sound devices specific to the platform
* The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
* devices to enable corresponding mixer paths
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 7849644..73bc779 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -175,6 +175,34 @@
SND_DEVICE_OUT_VOICE_SPEAKER,
};
+static const snd_device_t tasha_fluid_variant_devices[] = {
+ SND_DEVICE_OUT_SPEAKER,
+ SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+ SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+ SND_DEVICE_OUT_VOICE_SPEAKER,
+ SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+ SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,
+ SND_DEVICE_OUT_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+};
+
+static const snd_device_t tasha_liquid_variant_devices[] = {
+ SND_DEVICE_OUT_SPEAKER,
+ SND_DEVICE_OUT_SPEAKER_EXTERNAL_1,
+ SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1,
+ SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+ SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+ SND_DEVICE_IN_SPEAKER_MIC,
+ SND_DEVICE_IN_HEADSET_MIC,
+ SND_DEVICE_IN_VOICE_DMIC,
+ SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
+ SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
+ SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
+ SND_DEVICE_IN_QUAD_MIC,
+ SND_DEVICE_IN_HANDSET_STEREO_DMIC,
+ SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
+};
+
static void update_hardware_info_8084(struct hardware_info *hw_info, const char *snd_card_name)
{
if (!strcmp(snd_card_name, "apq8084-taiko-mtp-snd-card") ||
@@ -270,18 +298,24 @@
hw_info->snd_devices = (snd_device_t *)tomtom_8996_CDP_variant_devices;
hw_info->num_snd_devices = ARRAY_SIZE(tomtom_8996_CDP_variant_devices);
strlcpy(hw_info->dev_extn, "-cdp", sizeof(hw_info->dev_extn));
- } else if (!strcmp(snd_card_name, "msm8996-tomtom-stp-snd-card")) {
- strlcpy(hw_info->type, " stp", sizeof(hw_info->type));
- strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
- hw_info->snd_devices = (snd_device_t *)tomtom_stp_variant_devices;
- hw_info->num_snd_devices = ARRAY_SIZE(tomtom_stp_variant_devices);
- strlcpy(hw_info->dev_extn, "-stp", sizeof(hw_info->dev_extn));
} else if (!strcmp(snd_card_name, "msm8996-tomtom-liquid-snd-card")) {
strlcpy(hw_info->type, " liquid", sizeof(hw_info->type));
strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
hw_info->snd_devices = (snd_device_t *)tomtom_liquid_variant_devices;
hw_info->num_snd_devices = ARRAY_SIZE(tomtom_liquid_variant_devices);
strlcpy(hw_info->dev_extn, "-liquid", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8996-tasha-fluid-snd-card")) {
+ strlcpy(hw_info->type, " fluid", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
+ hw_info->snd_devices = (snd_device_t *)tasha_fluid_variant_devices;
+ hw_info->num_snd_devices = ARRAY_SIZE(tasha_fluid_variant_devices);
+ strlcpy(hw_info->dev_extn, "-fluid", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8996-tasha-liquid-snd-card")) {
+ strlcpy(hw_info->type, " liquid", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
+ hw_info->snd_devices = (snd_device_t *)tasha_liquid_variant_devices;
+ hw_info->num_snd_devices = ARRAY_SIZE(tasha_liquid_variant_devices);
+ strlcpy(hw_info->dev_extn, "-liquid", sizeof(hw_info->dev_extn));
} else if (!strcmp(snd_card_name, "msm8996-tasha-db-snd-card")) {
strlcpy(hw_info->type, " dragon-board", sizeof(hw_info->type));
strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 875cc2c..e617c0e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1040,40 +1040,55 @@
void *buff;
};
-static int send_codec_cal(acdb_loader_get_calibration_t acdb_loader_get_calibration, int fd)
+static void send_codec_cal(acdb_loader_get_calibration_t acdb_loader_get_calibration, int fd)
{
- int ret = 0, type;
+ int type;
for (type = WCD9XXX_ANC_CAL; type < WCD9XXX_MAX_CAL; type++) {
struct wcdcal_ioctl_buffer codec_buffer;
struct param_data calib;
+ int ret;
+ ret = 0;
calib.get_size = 1;
- ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
- &calib);
+ ret = acdb_loader_get_calibration(cal_name_info[type],
+ sizeof(struct param_data),
+ &calib);
if (ret < 0) {
- ALOGE("%s get_calibration failed\n", __func__);
- return ret;
+ ALOGE("%s: %s get_calibration size failed, err = %d\n",
+ __func__, cal_name_info[type], ret);
+ continue;
}
+
calib.get_size = 0;
calib.buff = malloc(calib.buff_size);
+ if (!calib.buff) {
+ ALOGE("%s: %s: No Memory for size = %d\n",
+ __func__, cal_name_info[type], calib.buff_size);
+ continue;
+ }
+
ret = acdb_loader_get_calibration(cal_name_info[type],
sizeof(struct param_data), &calib);
if (ret < 0) {
- ALOGE("%s get_calibration failed\n", __func__);
+ ALOGE("%s: %s get_calibration failed, err = %d\n",
+ __func__, cal_name_info[type], ret);
free(calib.buff);
- return ret;
+ continue;
}
+
codec_buffer.buffer = calib.buff;
codec_buffer.size = calib.data_size;
codec_buffer.cal_type = type;
if (ioctl(fd, SNDRV_CTL_IOCTL_HWDEP_CAL_TYPE, &codec_buffer) < 0)
- ALOGE("Failed to call ioctl for %s err=%d",
- cal_name_info[type], errno);
- ALOGD("%s cal sent for %s", __func__, cal_name_info[type]);
+ ALOGE("%s: %s Failed to call ioctl, err=%d",
+ __func__, cal_name_info[type], errno);
+ else
+ ALOGD("%s: %s cal sent successfully\n",
+ __func__, cal_name_info[type]);
+
free(calib.buff);
}
- return ret;
}
static void audio_hwdep_send_cal(struct platform_data *plat_data)
@@ -1094,8 +1109,8 @@
dlerror());
return;
}
- if (send_codec_cal(acdb_loader_get_calibration, fd) < 0)
- ALOGE("%s: Could not send anc cal", __FUNCTION__);
+
+ send_codec_cal(acdb_loader_get_calibration, fd);
}
static int platform_acdb_init(void *platform)
@@ -1827,11 +1842,10 @@
int snd_device = SND_DEVICE_OUT_SPEAKER;
if (usecase->type == PCM_PLAYBACK)
- snd_device = platform_get_output_snd_device(adev->platform,
- usecase->stream.out);
+ snd_device = usecase->out_snd_device;
else if ((usecase->type == PCM_HFP_CALL) || (usecase->type == PCM_CAPTURE))
- snd_device = platform_get_input_snd_device(adev->platform,
- adev->primary_output->devices);
+ snd_device = usecase->in_snd_device;
+
acdb_dev_id = acdb_device_table[audio_extn_get_spkr_prot_snd_device(snd_device)];
if (acdb_dev_id < 0) {
ALOGE("%s: Could not find acdb id for device(%d)",
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 2247933..fa9f4bb 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -45,6 +45,15 @@
(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)
+/*
+ * Below are the input devices for which back end is same, SLIMBUS_0_TX.
+ * All these devices are handled by the internal HW codec. We can
+ * enable any one of these devices at any time
+ */
+#define AUDIO_DEVICE_IN_ALL_CODEC_BACKEND \
+ (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC | \
+ AUDIO_DEVICE_IN_WIRED_HEADSET | AUDIO_DEVICE_IN_VOICE_CALL) & ~AUDIO_DEVICE_BIT_IN
+
/* Sound devices specific to the platform
* The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
* devices to enable corresponding mixer paths
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 2748568..3874f0b 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -56,6 +56,15 @@
ALOGV("%s: ctxt %p, strength: %d", __func__, context, strength);
context->strength = strength;
+ /*
+ * Zero strength is not equivalent to disable state as down mix
+ * is still happening for multichannel inputs.
+ * For better user experience, explicitly disable virtualizer module
+ * when strength is 0.
+ */
+ offload_virtualizer_set_enable_flag(&(context->offload_virt),
+ ((strength > 0) && !(context->temp_disabled)) ?
+ true : false);
offload_virtualizer_set_strength(&(context->offload_virt), strength);
if (context->ctl)
offload_virtualizer_send_params(context->ctl, &context->offload_virt,