Merge "hal: add a new key to query for decoder viability"
diff --git a/hal/Android.mk b/hal/Android.mk
index d7b7c50..f83faa5 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -231,10 +231,6 @@
LOCAL_SRC_FILES += audio_extn/source_track.c
endif
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUDIOSPHERE)),true)
- LOCAL_CFLAGS += -DAUDIOSPHERE_ENABLED
-endif
-
LOCAL_SHARED_LIBRARIES := \
liblog \
libcutils \
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 34da4fe..061af81 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -81,8 +81,6 @@
/* Query offload playback instances count */
#define AUDIO_PARAMETER_OFFLOAD_NUM_ACTIVE "offload_num_active"
#define AUDIO_PARAMETER_HPX "HPX"
-#define AUDIO_PARAMETER_KEY_ASPHERE_ENABLE "asphere_enable"
-#define AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH "asphere_strength"
#ifndef FM_ENABLED
#define audio_extn_fm_set_parameters(adev, parms) (0)
@@ -558,103 +556,6 @@
return ret;
}
-#ifndef AUDIOSPHERE_ENABLED
-#define audio_extn_asphere_set_parameters(adev, parms) (0)
-#define audio_extn_asphere_get_parameters(adev, query, reply) (0)
-#else
-int32_t audio_extn_asphere_set_parameters(const struct audio_device *adev,
- struct str_parms *parms)
-{
- int ret = 0, val[2];
- char value[32] = {0};
- int set_enable, set_strength;
- int enable = -1, strength = -1;
- struct mixer_ctl *ctl = NULL;
- const char *mixer_ctl_name = "MSM ASphere Set Param";
- char propValue[PROPERTY_VALUE_MAX] = {0};
- bool asphere_prop_enabled = false;
-
- if (property_get("audio.pp.asphere.enabled", propValue, "false")) {
- if (!strncmp("true", propValue, 4))
- asphere_prop_enabled = true;
- }
-
- if (!asphere_prop_enabled) {
- ALOGV("%s: property not set!!! not doing anything", __func__);
- return ret;
- }
-
- set_enable = str_parms_get_str(parms,
- AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- value, sizeof(value));
- if (set_enable > 0)
- enable = atoi(value);
-
- set_strength = str_parms_get_str(parms,
- AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- value, sizeof(value));
- if (set_strength > 0)
- strength = atoi(value);
-
- if (set_enable >= 0 || set_strength >= 0) {
- 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;
- }
- ALOGD("%s: set ctl \"%s:%d,%d\"",
- __func__, mixer_ctl_name, enable, strength);
- val[0] = enable;
- val[1] = strength;
- ret = mixer_ctl_set_array(ctl, val, sizeof(val)/sizeof(val[0]));
- if (ret)
- ALOGE("%s: set ctl failed!!!\"%s:%d,%d\"",
- __func__, mixer_ctl_name, enable, strength);
- }
- ALOGV("%s: exit ret %d", __func__, ret);
- return ret;
-}
-
-int32_t audio_extn_asphere_get_parameters(const struct audio_device *adev,
- struct str_parms *query,
- struct str_parms *reply)
-{
- int ret = 0, val[2] = {-1, -1};
- char value[32] = {0};
- int get_enable, get_strength;
- struct mixer_ctl *ctl = NULL;
- const char *mixer_ctl_name = "MSM ASphere Set Param";
-
- get_enable = str_parms_get_str(query,
- AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- value, sizeof(value));
- get_strength = str_parms_get_str(query,
- AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- value, sizeof(value));
- if (get_enable > 0 || get_strength > 0) {
- 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;
- }
- ret = mixer_ctl_get_array(ctl, val, sizeof(val)/sizeof(val[0]));
- if (ret)
- ALOGE("%s: got ctl failed!!! \"%s:%d,%d\"",
- __func__, mixer_ctl_name, val[0], val[1]);
- if (get_enable > 0)
- str_parms_add_int(reply,
- AUDIO_PARAMETER_KEY_ASPHERE_ENABLE, val[0]);
- if (get_strength > 0)
- str_parms_add_int(reply,
- AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH, val[1]);
- }
- ALOGV("%s: exit ret %d", __func__, ret);
- return ret;
-}
-#endif
-
void audio_extn_set_parameters(struct audio_device *adev,
struct str_parms *parms)
{
@@ -672,7 +573,8 @@
audio_extn_hpx_set_parameters(adev, parms);
audio_extn_pm_set_parameters(parms);
audio_extn_source_track_set_parameters(adev, parms);
- audio_extn_asphere_set_parameters(adev, parms);
+ if (adev->offload_effects_set_parameters != NULL)
+ adev->offload_effects_set_parameters(parms);
}
void audio_extn_get_parameters(const struct audio_device *adev,
@@ -686,7 +588,8 @@
audio_extn_dts_eagle_get_parameters(adev, query, reply);
audio_extn_hpx_get_parameters(query, reply);
audio_extn_source_track_get_parameters(adev, query, reply);
- audio_extn_asphere_get_parameters(adev, query, reply);
+ if (adev->offload_effects_get_parameters != NULL)
+ adev->offload_effects_get_parameters(query, reply);
kv_pairs = str_parms_to_str(reply);
ALOGD_IF(kv_pairs != NULL, "%s: returns %s", __func__, kv_pairs);
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index a0588a3..a73dfa1 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -1,5 +1,5 @@
/* hfp.c
-Copyright (c) 2012-2014, The Linux Foundation. All rights reserved.
+Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
@@ -271,7 +271,7 @@
}
/* 2. Disable echo reference while stopping hfp */
- platform_set_echo_reference(adev->platform, false);
+ platform_set_echo_reference(adev, false, uc_info->devices);
/* 3. Get and set stream specific mixer controls */
disable_audio_route(adev, uc_info);
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 37125f7..512a584 100755
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -177,6 +177,7 @@
[USECASE_AUDIO_PLAYBACK_OFFLOAD8] = "compress-offload-playback8",
[USECASE_AUDIO_PLAYBACK_OFFLOAD9] = "compress-offload-playback9",
#endif
+ [USECASE_AUDIO_PLAYBACK_ULL] = "audio-ull-playback",
[USECASE_AUDIO_RECORD] = "audio-record",
[USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
[USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
@@ -900,7 +901,7 @@
adev->active_input->source == AUDIO_SOURCE_MIC)) &&
adev->primary_output && !adev->primary_output->standby) {
out_device = adev->primary_output->devices;
- platform_set_echo_reference(adev->platform, false);
+ platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
} else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
}
@@ -1139,7 +1140,10 @@
}
audio_extn_perf_lock_release();
+ ALOGV("%s: pcm_prepare start", __func__);
+ pcm_prepare(in->pcm);
ALOGV("%s: exit", __func__);
+
return ret;
error_open:
@@ -1607,8 +1611,14 @@
}
break;
}
+
platform_set_stream_channel_map(adev->platform, out->channel_mask,
out->pcm_device_id);
+
+ ALOGV("%s: pcm_prepare start", __func__);
+ if (pcm_is_ready(out->pcm))
+ pcm_prepare(out->pcm);
+
} else {
platform_set_stream_channel_map(adev->platform, out->channel_mask,
out->pcm_device_id);
@@ -1651,7 +1661,9 @@
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
+
ALOGV("%s: exit", __func__);
+
return 0;
error_open:
stop_output_stream(out);
@@ -1787,6 +1799,9 @@
pthread_mutex_lock(&out->lock);
if (!out->standby) {
+ if (adev->adm_deregister_stream)
+ adev->adm_deregister_stream(adev->adm_data, out->handle);
+
pthread_mutex_lock(&adev->lock);
out->standby = true;
if (!is_offload_usecase(out->usecase)) {
@@ -2139,6 +2154,8 @@
out->standby = true;
goto exit;
}
+ if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD && adev->adm_register_output_stream)
+ adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags);
}
if (adev->mChannelStatusSet == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
@@ -2184,15 +2201,24 @@
if (out->pcm) {
if (out->muted)
memset((void *)buffer, 0, bytes);
+
ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
+
+ if (adev->adm_request_focus)
+ adev->adm_request_focus(adev->adm_data, out->handle);
+
if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY)
ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
else
ret = pcm_write(out->pcm, (void *)buffer, bytes);
+
if (ret < 0)
ret = -errno;
else if (ret == 0)
out->written += bytes / (out->config.channels * sizeof(short));
+
+ if (adev->adm_abandon_focus)
+ adev->adm_abandon_focus(adev->adm_data, out->handle);
}
}
@@ -2508,6 +2534,9 @@
}
if (!in->standby) {
+ if (adev->adm_deregister_stream)
+ adev->adm_deregister_stream(adev->adm_data, in->capture_handle);
+
pthread_mutex_lock(&adev->lock);
in->standby = true;
if (in->pcm) {
@@ -2653,8 +2682,13 @@
goto exit;
}
in->standby = 0;
+ if (adev->adm_register_input_stream)
+ adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags);
}
+ if (adev->adm_request_focus)
+ adev->adm_request_focus(adev->adm_data, in->capture_handle);
+
if (in->pcm) {
if (audio_extn_ssr_get_enabled() &&
audio_channel_count_from_in_mask(in->channel_mask) == 6)
@@ -2669,6 +2703,9 @@
ret = -errno;
}
+ if (adev->adm_abandon_focus)
+ adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+
/*
* Instead of writing zeroes here, we could trust the hardware
* to always provide zeroes when muted.
@@ -2988,6 +3025,10 @@
out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
out->config = pcm_config_afe_proxy_playback;
adev->voice_tx_output = out;
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
+ out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
+ out->config = pcm_config_low_latency;
+ out->sample_rate = out->config.rate;
} else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
format = AUDIO_FORMAT_PCM_16_BIT;
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
@@ -3457,6 +3498,7 @@
in->standby = 1;
in->channel_mask = config->channel_mask;
in->capture_handle = handle;
+ in->flags = flags;
/* Update config params with the requested sample rate and channels */
in->usecase = USECASE_AUDIO_RECORD;
@@ -3552,7 +3594,7 @@
ALOGD("%s: enter:stream_handle(%p)",__func__, in);
/* Disable echo reference while closing input stream */
- platform_set_echo_reference(adev->platform, false);
+ platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
@@ -3599,10 +3641,13 @@
audio_route_free(adev->audio_route);
free(adev->snd_dev_ref_cnt);
platform_deinit(adev->platform);
+ if (adev->adm_deinit)
+ adev->adm_deinit(adev->adm_data);
free(device);
adev = NULL;
}
pthread_mutex_unlock(&adev_init_lock);
+
return 0;
}
@@ -3614,6 +3659,7 @@
{
switch (period_size) {
case 160:
+ case 192:
case 240:
case 320:
case 480:
@@ -3737,6 +3783,36 @@
adev->offload_effects_set_hpx_state =
(int (*)(bool))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_set_hpx_state");
+ adev->offload_effects_get_parameters =
+ (void (*)(struct str_parms *, struct str_parms *))
+ dlsym(adev->offload_effects_lib,
+ "offload_effects_bundle_get_parameters");
+ adev->offload_effects_set_parameters =
+ (void (*)(struct str_parms *))dlsym(adev->offload_effects_lib,
+ "offload_effects_bundle_set_parameters");
+ }
+ }
+
+ if (access(ADM_LIBRARY_PATH, R_OK) == 0) {
+ adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW);
+ if (adev->adm_lib == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, ADM_LIBRARY_PATH);
+ } else {
+ ALOGV("%s: DLOPEN successful for %s", __func__, ADM_LIBRARY_PATH);
+ adev->adm_init = (adm_init_t)
+ dlsym(adev->adm_lib, "adm_init");
+ adev->adm_deinit = (adm_deinit_t)
+ dlsym(adev->adm_lib, "adm_deinit");
+ adev->adm_register_input_stream = (adm_register_input_stream_t)
+ dlsym(adev->adm_lib, "adm_register_input_stream");
+ adev->adm_register_output_stream = (adm_register_output_stream_t)
+ dlsym(adev->adm_lib, "adm_register_output_stream");
+ adev->adm_deregister_stream = (adm_deregister_stream_t)
+ dlsym(adev->adm_lib, "adm_deregister_stream");
+ adev->adm_request_focus = (adm_request_focus_t)
+ dlsym(adev->adm_lib, "adm_request_focus");
+ adev->adm_abandon_focus = (adm_abandon_focus_t)
+ dlsym(adev->adm_lib, "adm_abandon_focus");
}
}
@@ -3770,6 +3846,9 @@
pthread_mutex_unlock(&adev_init_lock);
+ if (adev->adm_init)
+ adev->adm_data = adev->adm_init();
+
ALOGV("%s: exit", __func__);
return 0;
}
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 45e90b7..983a89e 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -49,6 +49,7 @@
#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"
+#define ADM_LIBRARY_PATH "/system/vendor/lib/libadm.so"
/* Flags used to initialize acdb_settings variable that goes to ACDB library */
#define NONE_FLAG 0x00000000
@@ -93,6 +94,7 @@
USECASE_AUDIO_PLAYBACK_OFFLOAD8,
USECASE_AUDIO_PLAYBACK_OFFLOAD9,
#endif
+ USECASE_AUDIO_PLAYBACK_ULL,
/* FM usecase */
USECASE_AUDIO_PLAYBACK_FM,
@@ -233,6 +235,7 @@
bool enable_ns;
audio_format_t format;
audio_io_handle_t capture_handle;
+ audio_input_flags_t flags;
bool is_st_session;
bool is_st_session_active;
@@ -285,6 +288,14 @@
struct stream_app_type_cfg app_type_cfg;
};
+typedef void* (*adm_init_t)();
+typedef void (*adm_deinit_t)(void *);
+typedef void (*adm_register_output_stream_t)(void *, audio_io_handle_t, audio_output_flags_t);
+typedef void (*adm_register_input_stream_t)(void *, audio_io_handle_t, audio_input_flags_t);
+typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
+typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
+typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
+
struct audio_device {
struct audio_hw_device device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
@@ -323,6 +334,19 @@
struct sound_card_status snd_card_status;
int (*offload_effects_set_hpx_state)(bool);
+ void (*offload_effects_get_parameters)(struct str_parms *,
+ struct str_parms *);
+ void (*offload_effects_set_parameters)(struct str_parms *);
+
+ void *adm_data;
+ void *adm_lib;
+ adm_init_t adm_init;
+ adm_deinit_t adm_deinit;
+ adm_register_input_stream_t adm_register_input_stream;
+ adm_register_output_stream_t adm_register_output_stream;
+ adm_deregister_stream_t adm_deregister_stream;
+ adm_request_focus_t adm_request_focus;
+ adm_abandon_focus_t adm_abandon_focus;
};
int select_devices(struct audio_device *adev,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index fc7d038..63adaa1 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -247,6 +247,7 @@
void *edid_info;
bool edid_valid;
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
+ char ec_ref_mixer_path[64];
};
static bool is_external_codec = false;
@@ -280,6 +281,7 @@
[USECASE_AUDIO_PLAYBACK_OFFLOAD8] = {-1, -1},
[USECASE_AUDIO_PLAYBACK_OFFLOAD9] = {-1, -1},
#endif
+ [USECASE_AUDIO_PLAYBACK_ULL] = {MULTIMEDIA3_PCM_DEVICE, MULTIMEDIA3_PCM_DEVICE},
[USECASE_AUDIO_RECORD] = {AUDIO_RECORD_PCM_DEVICE, AUDIO_RECORD_PCM_DEVICE},
[USECASE_AUDIO_RECORD_COMPRESS] = {COMPRESS_CAPTURE_DEVICE, COMPRESS_CAPTURE_DEVICE},
[USECASE_AUDIO_RECORD_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
@@ -663,6 +665,7 @@
{TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_OFFLOAD8)},
{TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_OFFLOAD9)},
#endif
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_ULL)},
{TO_NAME_INDEX(USECASE_AUDIO_RECORD)},
{TO_NAME_INDEX(USECASE_AUDIO_RECORD_LOW_LATENCY)},
{TO_NAME_INDEX(USECASE_VOICE_CALL)},
@@ -968,6 +971,41 @@
}
}
+void platform_set_echo_reference(struct audio_device *adev, bool enable,
+ audio_devices_t out_device)
+{
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+ snd_device_t snd_device = SND_DEVICE_NONE;
+ struct stream_out out;
+
+ out.devices = out_device;
+
+ if (strcmp(my_data->ec_ref_mixer_path, "")) {
+ ALOGV("%s: disabling %s", __func__, my_data->ec_ref_mixer_path);
+ audio_route_reset_and_update_path(adev->audio_route,
+ my_data->ec_ref_mixer_path);
+ }
+
+ if (enable) {
+ snd_device = platform_get_output_snd_device(adev->platform, &out);
+
+ if (adev->snd_dev_ref_cnt[SND_DEVICE_OUT_HEADPHONES_44_1] > 0)
+ strlcpy(my_data->ec_ref_mixer_path, "echo-reference headphones-44.1",
+ sizeof(my_data->ec_ref_mixer_path));
+ else if ((snd_device == SND_DEVICE_OUT_SPEAKER_VBAT) ||
+ (snd_device == SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT))
+ strlcpy(my_data->ec_ref_mixer_path, "vbat-speaker echo-reference",
+ sizeof(my_data->ec_ref_mixer_path));
+ else
+ strlcpy(my_data->ec_ref_mixer_path, "echo-reference",
+ sizeof(my_data->ec_ref_mixer_path));
+
+
+ ALOGD("%s: enabling %s", __func__, my_data->ec_ref_mixer_path);
+ audio_route_apply_and_update_path(adev->audio_route,
+ my_data->ec_ref_mixer_path);
+ }
+}
void platform_set_gsm_mode(void *platform, bool enable)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -986,27 +1024,7 @@
}
}
-void platform_set_echo_reference(void *platform, bool enable)
-{
- struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
- char *mixer_path_name = "echo-reference";
- if(my_data->is_vbat_speaker)
- mixer_path_name = "vbat-speaker echo-reference";
-
- if (my_data->ec_ref_enabled) {
- my_data->ec_ref_enabled = false;
- ALOGV("%s: disabling echo-reference", __func__);
- audio_route_reset_and_update_path(adev->audio_route, mixer_path_name);
- }
-
- if (enable) {
- my_data->ec_ref_enabled = true;
- ALOGD("%s: enabling echo-reference", __func__);
- audio_route_apply_and_update_path(adev->audio_route, mixer_path_name);
- }
-}
static struct csd_data *open_csd_client()
{
@@ -1283,6 +1301,10 @@
ALOGE("%s error in sending vbat adc data to acdb", __func__);
}
+ /* MAD calibration is handled by sound trigger HAL, skip here */
+ if (type == WCD9XXX_MAD_CAL)
+ continue;
+
calib.get_size = 1;
ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
&calib);
@@ -2679,7 +2701,7 @@
my_data->fluence_in_voice_call == false) {
snd_device = SND_DEVICE_IN_HANDSET_MIC;
if (audio_extn_hfp_is_active(adev))
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else {
snd_device = SND_DEVICE_IN_VOICE_DMIC;
adev->acdb_settings |= DMIC_FLAG;
@@ -2687,7 +2709,7 @@
} else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
if (audio_extn_hfp_is_active(adev))
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
if (adev->bt_wb_speech_enabled) {
if (adev->bluetooth_nrec)
@@ -2717,7 +2739,7 @@
} else {
snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
if (audio_extn_hfp_is_active(adev))
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
}
} else if (out_device & AUDIO_DEVICE_OUT_TELEPHONY_TX)
snd_device = SND_DEVICE_IN_VOICE_RX;
@@ -2775,7 +2797,7 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
}
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (my_data->fluence_type != FLUENCE_NONE &&
adev->active_input->enable_aec) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
@@ -2800,7 +2822,7 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
}
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (my_data->fluence_type != FLUENCE_NONE &&
adev->active_input->enable_ns) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
@@ -2825,9 +2847,9 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
}
- platform_set_echo_reference(adev->platform,false);
+ platform_set_echo_reference(adev, false, out_device);
} else
- platform_set_echo_reference(adev->platform, false);
+ platform_set_echo_reference(adev, false, out_device);
}
} else if (source == AUDIO_SOURCE_MIC) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC &&
@@ -2835,10 +2857,10 @@
if(my_data->fluence_in_audio_rec) {
if(my_data->fluence_type & FLUENCE_QUAD_MIC) {
snd_device = SND_DEVICE_IN_HANDSET_QMIC;
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
snd_device = SND_DEVICE_IN_HANDSET_DMIC;
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
}
}
}
@@ -3931,9 +3953,11 @@
ALOGI("%s Codec selected backend: %d current bit width: %d and sample rate: %d",
__func__, backend_idx, bit_width, sample_rate);
- // For voice calls use default configuration
+ // For voice calls use default configuration i.e. 16b/48K, only applicable to
+ // default backend
// force routing is not required here, caller will do it anyway
- if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
+ if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ backend_idx == DEFAULT_CODEC_BACKEND) {
ALOGW("%s:Use default bw and sr for voice/voip calls ",__func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index bf5e834..5d0ad2d 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -236,6 +236,7 @@
#define DEEP_BUFFER_PCM_DEVICE 0
#define AUDIO_RECORD_PCM_DEVICE 0
#define MULTIMEDIA2_PCM_DEVICE 1
+#define MULTIMEDIA3_PCM_DEVICE 4
#define FM_PLAYBACK_PCM_DEVICE 5
#define FM_CAPTURE_PCM_DEVICE 6
#define HFP_PCM_RX 5
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 175a3a6..d519764 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -226,6 +226,7 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
+ char ec_ref_mixer_path[64];
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
};
@@ -235,7 +236,7 @@
[USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
LOWLATENCY_PCM_DEVICE},
[USECASE_AUDIO_PLAYBACK_MULTI_CH] = {MULTIMEDIA2_PCM_DEVICE,
- MULTIMEDIA2_PCM_DEVICE},
+ MULTIMEDIA2_PCM_DEVICE},
[USECASE_AUDIO_PLAYBACK_OFFLOAD] =
{PLAYBACK_OFFLOAD_DEVICE, PLAYBACK_OFFLOAD_DEVICE},
#ifdef MULTIPLE_OFFLOAD_ENABLED
@@ -256,6 +257,9 @@
[USECASE_AUDIO_PLAYBACK_OFFLOAD9] =
{PLAYBACK_OFFLOAD_DEVICE9, PLAYBACK_OFFLOAD_DEVICE9},
#endif
+ [USECASE_AUDIO_PLAYBACK_ULL] = {MULTIMEDIA3_PCM_DEVICE,
+ MULTIMEDIA3_PCM_DEVICE},
+
[USECASE_AUDIO_RECORD] = {AUDIO_RECORD_PCM_DEVICE, AUDIO_RECORD_PCM_DEVICE},
[USECASE_AUDIO_RECORD_COMPRESS] = {COMPRESS_CAPTURE_DEVICE, COMPRESS_CAPTURE_DEVICE},
[USECASE_AUDIO_RECORD_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
@@ -607,6 +611,7 @@
{TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_OFFLOAD8)},
{TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_OFFLOAD9)},
#endif
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_ULL)},
{TO_NAME_INDEX(USECASE_AUDIO_RECORD)},
{TO_NAME_INDEX(USECASE_AUDIO_RECORD_LOW_LATENCY)},
{TO_NAME_INDEX(USECASE_VOICE_CALL)},
@@ -710,23 +715,39 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
-void platform_set_echo_reference(void *platform, bool enable)
+void platform_set_echo_reference(struct audio_device *adev, bool enable,
+ audio_devices_t out_device)
{
- struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+ snd_device_t snd_device = SND_DEVICE_NONE;
+ struct stream_out out;
- if (my_data->ec_ref_enabled) {
- my_data->ec_ref_enabled = false;
- ALOGV("%s: disabling echo-reference", __func__);
- audio_route_reset_and_update_path(adev->audio_route, "echo-reference");
+ out.devices = out_device;
+
+ if (strcmp(my_data->ec_ref_mixer_path, "")) {
+ ALOGV("%s: disabling %s", __func__, my_data->ec_ref_mixer_path);
+ audio_route_reset_and_update_path(adev->audio_route,
+ my_data->ec_ref_mixer_path);
}
if (enable) {
- my_data->ec_ref_enabled = true;
- ALOGD("%s: enabling echo-reference", __func__);
- audio_route_apply_and_update_path(adev->audio_route, "echo-reference");
- }
+ strlcpy(my_data->ec_ref_mixer_path, "echo-reference",
+ sizeof(my_data->ec_ref_mixer_path));
+ snd_device = platform_get_output_snd_device(adev->platform, &out);
+ /*
+ * If native audio device reference count > 0, then apply codec EC otherwise
+ * fallback to headphones if so or default
+ */
+ if (adev->snd_dev_ref_cnt[SND_DEVICE_OUT_HEADPHONES_44_1] > 0)
+ platform_add_backend_name(my_data->ec_ref_mixer_path,
+ SND_DEVICE_OUT_HEADPHONES_44_1);
+ else
+ platform_add_backend_name(my_data->ec_ref_mixer_path, snd_device);
+ ALOGD("%s: enabling %s", __func__, my_data->ec_ref_mixer_path);
+ audio_route_apply_and_update_path(adev->audio_route,
+ my_data->ec_ref_mixer_path);
+ }
}
static struct csd_data *open_csd_client(bool i2s_ext_modem)
@@ -993,6 +1014,10 @@
struct wcdcal_ioctl_buffer codec_buffer;
struct param_data calib;
+ /* MAD calibration is handled by sound trigger HAL, skip here */
+ if (type == WCD9XXX_MAD_CAL)
+ continue;
+
calib.get_size = 1;
ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
&calib);
@@ -2270,14 +2295,14 @@
my_data->fluence_in_voice_call == false) {
snd_device = SND_DEVICE_IN_HANDSET_MIC;
if (audio_extn_hfp_is_active(adev))
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else {
snd_device = SND_DEVICE_IN_VOICE_DMIC;
}
} else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
if (audio_extn_hfp_is_active(adev))
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
if (adev->bt_wb_speech_enabled) {
if (adev->bluetooth_nrec)
@@ -2305,7 +2330,7 @@
} else {
snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
if (audio_extn_hfp_is_active(adev))
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
}
} else if (out_device & AUDIO_DEVICE_OUT_TELEPHONY_TX)
snd_device = SND_DEVICE_IN_VOICE_RX;
@@ -2353,7 +2378,7 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
}
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (adev->active_input->enable_aec) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
@@ -2375,7 +2400,7 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
}
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (adev->active_input->enable_ns) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
@@ -2397,9 +2422,9 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
}
- platform_set_echo_reference(adev->platform, false);
+ platform_set_echo_reference(adev, false, out_device);
} else
- platform_set_echo_reference(adev->platform, false);
+ platform_set_echo_reference(adev, false, out_device);
}
} else if (source == AUDIO_SOURCE_MIC) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC &&
@@ -2407,10 +2432,10 @@
if(my_data->fluence_in_audio_rec) {
if(my_data->fluence_type & FLUENCE_QUAD_MIC) {
snd_device = SND_DEVICE_IN_HANDSET_QMIC;
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
} else if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
snd_device = SND_DEVICE_IN_HANDSET_DMIC;
- platform_set_echo_reference(adev->platform, true);
+ platform_set_echo_reference(adev, true, out_device);
}
}
}
@@ -3404,9 +3429,12 @@
ALOGI("%s Codec selected backend: %d current bit width: %d and sample rate: %d",
__func__, backend_idx, bit_width, sample_rate);
- // For voice calls use default configuration
+
+ // For voice calls use default configuration i.e. 16b/48K, only applicable to
+ // default backend
// force routing is not required here, caller will do it anyway
- if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
+ if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ backend_idx == DEFAULT_CODEC_BACKEND) {
ALOGW("%s:Use default bw and sr for voice/voip calls ",__func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 79e0816..4787b86 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -217,6 +217,7 @@
#define DEEP_BUFFER_PCM_DEVICE 0
#define AUDIO_RECORD_PCM_DEVICE 0
#define MULTIMEDIA2_PCM_DEVICE 1
+#define MULTIMEDIA3_PCM_DEVICE 4
#define FM_PLAYBACK_PCM_DEVICE 5
#define FM_CAPTURE_PCM_DEVICE 6
#define HFP_PCM_RX 5
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 53ddb48..9430721 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -101,7 +101,7 @@
struct audio_usecase *usecase, snd_device_t snd_device);
int platform_get_usecase_index(const char * usecase);
int platform_set_usecase_pcm_id(audio_usecase_t usecase, int32_t type, int32_t pcm_id);
-void platform_set_echo_reference(void *platform, bool enable);
+void platform_set_echo_reference(struct audio_device *adev, bool enable, audio_devices_t out_device);
void platform_get_device_to_be_id_map(int **be_id_map, int *length);
int platform_set_channel_allocation(void *platform, int channel_alloc);
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index a638c1d..17b7135 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -907,6 +907,53 @@
mLimitRingtoneVolume = false;
}
}
+
+void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+
+ if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+ ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+ return;
+ }
+ bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ // Use reverse loop to make sure any low latency usecases (generally tones)
+ // are not routed before non LL usecases (generally music).
+ // We can safely assume that LL output would always have lower index,
+ // and use this work-around to avoid routing of output with music stream
+ // from the context of short lived LL output.
+ // Note: in case output's share backend(HAL sharing is implicit) all outputs
+ // gets routing update while processing first output itself.
+ for (size_t i = mOutputs.size(); i > 0; i--) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
+ audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || outputDesc != mPrimaryOutput) {
+ setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(outputDesc, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0) {
+ setInputDevice(activeInput, getNewInputDevice(activeInput));
+ }
+
+}
+
status_t AudioPolicyManagerCustom::stopSource(sp<SwAudioOutputDescriptor> outputDesc,
audio_stream_type_t stream,
bool forceDeviceUpdate)
@@ -1816,6 +1863,32 @@
return status;
}
+void AudioPolicyManagerCustom::closeAllInputs() {
+ bool patchRemoved = false;
+
+ for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
+ ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (patch_index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(patch_index);
+ patchRemoved = true;
+ }
+ if ((inputDesc->mIsSoundTrigger) && (mInputs.size() == 1)) {
+ ALOGD("Do not close sound trigger input handle");
+ } else {
+ mpClientInterface->closeInput(mInputs.keyAt(input_index));
+ mInputs.removeItem(mInputs.keyAt(input_index));
+ }
+ }
+ nextAudioPortGeneration();
+
+ if (patchRemoved) {
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+}
+
AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface),
mHdmiAudioDisabled(false),
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 53c9a1b..66f9c38 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -54,6 +54,8 @@
const char *device_address,
const char *device_name);
virtual void setPhoneState(audio_mode_t state);
+ virtual void setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
@@ -73,6 +75,7 @@
// indicates to the audio policy manager that the input stops being used.
virtual status_t stopInput(audio_io_handle_t input,
audio_session_t session);
+ virtual void closeAllInputs();
protected:
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index ed46f17..3b9787c 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -21,6 +21,11 @@
LOCAL_SRC_FILES += hw_accelerator.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUDIOSPHERE)),true)
+ LOCAL_CFLAGS += -DAUDIOSPHERE_ENABLED
+ LOCAL_SRC_FILES += asphere.c
+endif
+
LOCAL_CFLAGS+= -O2 -fvisibility=hidden
ifneq ($(strip $(AUDIO_FEATURE_DISABLED_DTS_EAGLE)),true)
diff --git a/post_proc/asphere.c b/post_proc/asphere.c
new file mode 100644
index 0000000..bbf1056
--- /dev/null
+++ b/post_proc/asphere.c
@@ -0,0 +1,301 @@
+/* Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#define LOG_TAG "audio_pp_asphere"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdbool.h>
+#include <sys/stat.h>
+#include <cutils/log.h>
+#include <cutils/list.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+#include <hardware/audio_effect.h>
+#include "bundle.h"
+#include "equalizer.h"
+#include "bass_boost.h"
+#include "virtualizer.h"
+#include "reverb.h"
+#include "asphere.h"
+
+#define ASPHERE_MIXER_NAME "MSM ASphere Set Param"
+
+#define AUDIO_PARAMETER_KEY_ASPHERE_STATUS "asphere_status"
+#define AUDIO_PARAMETER_KEY_ASPHERE_ENABLE "asphere_enable"
+#define AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH "asphere_strength"
+
+#define AUDIO_ASPHERE_EVENT_NODE "/data/misc/audio_pp/event_node"
+
+enum {
+ ASPHERE_ACTIVE = 0,
+ ASPHERE_SUSPENDED,
+ ASPHERE_ERROR
+};
+
+struct asphere_module {
+ bool enabled;
+ int status;
+ int strength;
+ pthread_mutex_t lock;
+ int init_status;
+};
+
+static struct asphere_module asphere;
+pthread_once_t asphere_once = PTHREAD_ONCE_INIT;
+
+static int asphere_create_app_notification_node(void)
+{
+ int fd;
+ if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_CREAT|O_TRUNC|O_WRONLY,
+ S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH)) < 0) {
+ ALOGE("creating notification node failed %d", errno);
+ return -EINVAL;
+ }
+ chmod(AUDIO_ASPHERE_EVENT_NODE, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
+ close(fd);
+ ALOGD("%s: successfully created notification node %s",
+ __func__, AUDIO_ASPHERE_EVENT_NODE);
+ return 0;
+}
+
+static int asphere_notify_app(void)
+{
+ int fd;
+ if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_TRUNC|O_WRONLY)) < 0) {
+ ALOGE("opening notification node failed %d", errno);
+ return -EINVAL;
+ }
+ close(fd);
+ ALOGD("%s: successfully opened notification node", __func__);
+ return 0;
+}
+
+static int asphere_get_values_from_mixer(void)
+{
+ int ret = 0, val[2] = {-1, -1};
+ struct mixer_ctl *ctl = NULL;
+ struct mixer *mixer = mixer_open(MIXER_CARD);
+ if (mixer)
+ ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
+ if (!ctl) {
+ ALOGE("%s: could not get ctl for mixer cmd - %s",
+ __func__, ASPHERE_MIXER_NAME);
+ return -EINVAL;
+ }
+ ret = mixer_ctl_get_array(ctl, val, sizeof(val)/sizeof(val[0]));
+ if (!ret) {
+ asphere.enabled = (val[0] == 0) ? false : true;
+ asphere.strength = val[1];
+ }
+ ALOGD("%s: returned %d, enabled:%d, strength:%d",
+ __func__, ret, val[0], val[1]);
+
+ return ret;
+}
+
+static int asphere_set_values_to_mixer(void)
+{
+ int ret = 0, val[2] = {-1, -1};
+ struct mixer_ctl *ctl = NULL;
+ struct mixer *mixer = mixer_open(MIXER_CARD);
+ if (mixer)
+ ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
+ if (!ctl) {
+ ALOGE("%s: could not get ctl for mixer cmd - %s",
+ __func__, ASPHERE_MIXER_NAME);
+ return -EINVAL;
+ }
+ val[0] = ((asphere.status == ASPHERE_ACTIVE) && asphere.enabled) ? 1 : 0;
+ val[1] = asphere.strength;
+
+ ret = mixer_ctl_set_array(ctl, val, sizeof(val)/sizeof(val[0]));
+ ALOGD("%s: returned %d, enabled:%d, strength:%d",
+ __func__, ret, val[0], val[1]);
+
+ return ret;
+}
+
+static void asphere_init_once() {
+ ALOGD("%s", __func__);
+ pthread_mutex_init(&asphere.lock, NULL);
+ asphere.init_status = 1;
+ asphere_get_values_from_mixer();
+ asphere_create_app_notification_node();
+}
+
+static int asphere_init() {
+ pthread_once(&asphere_once, asphere_init_once);
+ return asphere.init_status;
+}
+
+void asphere_set_parameters(struct str_parms *parms)
+{
+ int ret = 0;
+ bool enable = false;
+ int strength = -1;
+ char value[32] = {0};
+ char propValue[PROPERTY_VALUE_MAX] = {0};
+ bool set_enable = false, set_strength = false;
+
+ if (!property_get("audio.pp.asphere.enabled", propValue, "false") ||
+ (strncmp("true", propValue, 4) != 0)) {
+ ALOGV("%s: property not set!!! not doing anything", __func__);
+ return;
+ }
+ if (asphere_init() != 1) {
+ ALOGW("%s: init check failed!!!", __func__);
+ return;
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
+ value, sizeof(value));
+ if (ret > 0) {
+ enable = (atoi(value) == 1) ? true : false;
+ set_enable = true;
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
+ value, sizeof(value));
+ if (ret > 0) {
+ strength = atoi(value);
+ if (strength >= 0 && strength <= 1000)
+ set_strength = true;
+ }
+
+ if (set_enable || set_strength) {
+ pthread_mutex_lock(&asphere.lock);
+ asphere.enabled = set_enable ? enable : asphere.enabled;
+ asphere.strength = set_strength ? strength : asphere.strength;
+ ret = asphere_set_values_to_mixer();
+ pthread_mutex_unlock(&asphere.lock);
+ ALOGV("%s: exit ret %d", __func__, ret);
+ }
+}
+
+void asphere_get_parameters(struct str_parms *query,
+ struct str_parms *reply)
+{
+ char value[32] = {0};
+ char propValue[PROPERTY_VALUE_MAX] = {0};
+ int get_status, get_enable, get_strength, ret;
+
+ if (!property_get("audio.pp.asphere.enabled", propValue, "false") ||
+ (strncmp("true", propValue, 4) != 0)) {
+ ALOGV("%s: property not set!!! not doing anything", __func__);
+ return;
+ }
+ if (asphere_init() != 1) {
+ ALOGW("%s: init check failed!!!", __func__);
+ return;
+ }
+
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
+ value, sizeof(value));
+ if (ret >= 0) {
+ str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
+ asphere.status);
+ }
+
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
+ value, sizeof(value));
+ if (ret >= 0) {
+ str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
+ asphere.enabled ? 1 : 0);
+ }
+
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
+ value, sizeof(value));
+ if (ret >= 0) {
+ str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
+ asphere.strength);
+ }
+}
+
+static bool effect_needs_asphere_concurrency_handling(effect_context_t *context)
+{
+ if (memcmp(&context->desc->type,
+ &equalizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+ memcmp(&context->desc->type,
+ &bassboost_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+ memcmp(&context->desc->type,
+ &virtualizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+ memcmp(&context->desc->type,
+ &ins_preset_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
+ memcmp(&context->desc->type,
+ &ins_env_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0)
+ return true;
+
+ return false;
+}
+
+void handle_asphere_on_effect_enabled(bool enable,
+ effect_context_t *context,
+ struct listnode *created_effects)
+{
+ struct listnode *node;
+ char propValue[PROPERTY_VALUE_MAX] = {0};
+
+ ALOGV("%s: effect %0x", __func__, context->desc->type.timeLow);
+ if (!property_get("audio.pp.asphere.enabled", propValue, "false") ||
+ (strncmp("true", propValue, 4) != 0)) {
+ ALOGV("%s: property not set!!! not doing anything", __func__);
+ return;
+ }
+ if (asphere_init() != 1) {
+ ALOGW("%s: init check failed!!!", __func__);
+ return;
+ }
+
+ if (!effect_needs_asphere_concurrency_handling(context)) {
+ ALOGV("%s: effect %0x, do not need concurrency handling",
+ __func__, context->desc->type.timeLow);
+ return;
+ }
+
+ list_for_each(node, created_effects) {
+ effect_context_t *fx_ctxt = node_to_item(node,
+ effect_context_t,
+ effects_list_node);
+ if (fx_ctxt != NULL &&
+ effect_needs_asphere_concurrency_handling(fx_ctxt) == true &&
+ fx_ctxt != context && effect_is_active(fx_ctxt) == true) {
+ ALOGV("%s: found another effect %0x, skip processing %0x", __func__,
+ fx_ctxt->desc->type.timeLow, context->desc->type.timeLow);
+ return;
+ }
+ }
+ pthread_mutex_lock(&asphere.lock);
+ asphere.status = enable ? ASPHERE_SUSPENDED : ASPHERE_ACTIVE;
+ asphere_set_values_to_mixer();
+ asphere_notify_app();
+ pthread_mutex_unlock(&asphere.lock);
+}
diff --git a/post_proc/asphere.h b/post_proc/asphere.h
new file mode 100644
index 0000000..d0e6830
--- /dev/null
+++ b/post_proc/asphere.h
@@ -0,0 +1,50 @@
+/* Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+
+#ifndef OFFLOAD_ASPHERE_H_
+#define OFFLOAD_ASPHERE_H_
+
+#include <cutils/str_parms.h>
+#include <cutils/list.h>
+#include "bundle.h"
+
+#ifdef AUDIOSPHERE_ENABLED
+void asphere_get_parameters(struct str_parms *query,
+ struct str_parms *reply);
+void asphere_set_parameters(struct str_parms *reply);
+void handle_asphere_on_effect_enabled(bool enable,
+ effect_context_t *context,
+ struct listnode *created_effects);
+#else
+#define asphere_get_parameters(query, reply) (0)
+#define asphere_set_parameters(parms) (0)
+#define handle_asphere_on_effect_enabled(enable, context, created_effects) (0)
+#endif /* AUDIOSPHERE_ENABLED */
+
+#endif /* OFFLOAD_ASPHERE_H_ */
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index b33f2d1..8df93cb 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -50,6 +50,7 @@
#include "bass_boost.h"
#include "virtualizer.h"
#include "reverb.h"
+#include "asphere.h"
#ifdef DTS_EAGLE
#include "effect_util.h"
@@ -455,6 +456,24 @@
}
/*
+ * Effect Bundle Set and get param operations.
+ * currently only handles audio sphere scenario,
+ * but the interface itself can be utilized for any effect.
+ */
+__attribute__ ((visibility ("default")))
+void offload_effects_bundle_get_parameters(struct str_parms *query,
+ struct str_parms *reply)
+{
+ asphere_get_parameters(query, reply);
+}
+
+__attribute__ ((visibility ("default")))
+void offload_effects_bundle_set_parameters(struct str_parms *parms)
+{
+ asphere_set_parameters(parms);
+}
+
+/*
* Effect operations
*/
int set_config(effect_context_t *context, effect_config_t *config)
@@ -810,6 +829,7 @@
status = -ENOSYS;
goto exit;
}
+ handle_asphere_on_effect_enabled(true, context, &created_effects_list);
context->state = EFFECT_STATE_ACTIVE;
if (context->ops.enable)
context->ops.enable(context);
@@ -824,6 +844,7 @@
status = -ENOSYS;
goto exit;
}
+ handle_asphere_on_effect_enabled(false, context, &created_effects_list);
context->state = EFFECT_STATE_INITIALIZED;
if (context->ops.disable)
context->ops.disable(context);
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index d363b77..b2f0952 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -300,20 +300,40 @@
return false;
}
-int configure_proxy_capture(struct mixer *mixer, int value) {
- const char *proxy_ctl_name = "AFE_PCM_RX Audio Mixer MultiMedia4";
+int set_control(const char* name, struct mixer *mixer, int value) {
struct mixer_ctl *ctl;
+ ctl = mixer_get_ctl_by_name(mixer, name);
+ if (ctl == NULL) {
+ ALOGW("%s: could not get %s ctl", __func__, name);
+ return -EINVAL;
+ }
+ if (mixer_ctl_set_value(ctl, 0, value) != 0) {
+ ALOGW("%s: error setting value %d on %s ", __func__, value, name);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int configure_proxy_capture(struct mixer *mixer, int value) {
+ int retval = 0;
+
if (value && acdb_send_audio_cal)
acdb_send_audio_cal(AFE_PROXY_ACDB_ID, ACDB_DEV_TYPE_OUT);
- ctl = mixer_get_ctl_by_name(mixer, proxy_ctl_name);
- if (ctl == NULL) {
- ALOGW("%s: could not get %s ctl", __func__, proxy_ctl_name);
- return -EINVAL;
- }
- if (mixer_ctl_set_value(ctl, 0, value) != 0)
- ALOGW("%s: error setting value %d on %s ", __func__, value, proxy_ctl_name);
+ retval = set_control("AFE_PCM_RX Audio Mixer MultiMedia4", mixer, value);
+
+ if (retval != 0)
+ return retval;
+
+ // Extending visualizer to capture for compress2 path as well.
+ // for extending it to multiple offload either this needs to be extended
+ // or need to find better solution to enable only active offload sessions
+
+ retval = set_control("AFE_PCM_RX Audio Mixer MultiMedia7", mixer, value);
+ if (retval != 0)
+ return retval;
return 0;
}