Merge tag 'android-7.1.2_r5' of https://android.googlesource.com/platform/hardware/qcom/audio into HEAD
Android 7.1.2 Release 5 (N2G47J)
Change-Id: Ic8c1180a568dde687a79e580eb3db59f08d6061a
diff --git a/hal/Android.mk b/hal/Android.mk
index fc2be7d..f3d2798 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -10,7 +10,7 @@
ifneq ($(filter msm8960,$(TARGET_BOARD_PLATFORM)),)
LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="2"
endif
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msm8998,$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM = msm8974
ifneq ($(filter msm8974,$(TARGET_BOARD_PLATFORM)),)
@@ -41,6 +41,12 @@
LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
MULTIPLE_HW_VARIANTS_ENABLED := true
endif
+ifneq ($(filter msm8998,$(TARGET_BOARD_PLATFORM)),)
+ LOCAL_CFLAGS := -DPLATFORM_MSM8998
+ LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+ LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
+ MULTIPLE_HW_VARIANTS_ENABLED := true
+endif
endif
ifneq ($(filter msm8916 msm8909 msm8952,$(TARGET_BOARD_PLATFORM)),)
@@ -60,6 +66,7 @@
platform_info.c \
audio_extn/ext_speaker.c \
audio_extn/audio_extn.c \
+ audio_extn/utils.c \
$(AUDIO_PLATFORM)/platform.c
ifdef MULTIPLE_HW_VARIANTS_ENABLED
@@ -86,6 +93,12 @@
$(LOCAL_PATH)/voice_extn \
external/expat/lib
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SMART_PA_TFA_98XX)),true)
+ LOCAL_SHARED_LIBRARIES += libexTfa98xx
+ LOCAL_CFLAGS += -DSMART_PA_TFA_98XX_SUPPORTED
+ LOCAL_SRC_FILES += audio_extn/tfa_98xx.c
+endif
+
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_MULTI_VOICE_SESSIONS)),true)
LOCAL_CFLAGS += -DMULTI_VOICE_SESSION_ENABLED
LOCAL_SRC_FILES += voice_extn/voice_extn.c
@@ -121,7 +134,7 @@
LOCAL_SRC_FILES += audio_extn/dsm_feedback.c
endif
-ifneq ($(filter msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8998,$(TARGET_BOARD_PLATFORM)),)
# push codec/mad calibration to HW dep node
# applicable to msm8992/8994 or newer platforms
LOCAL_CFLAGS += -DHWDEP_CAL_ENABLED
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 5ce3093..f53c9f2 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -57,6 +57,8 @@
#define audio_extn_hfp_is_active(adev) (0)
#define audio_extn_hfp_get_usecase() (-1)
#define audio_extn_hfp_set_parameters(adev, params) (0)
+#define audio_extn_hfp_set_mic_mute(adev, state) (0)
+
#else
bool audio_extn_hfp_is_active(struct audio_device *adev);
@@ -64,6 +66,8 @@
void audio_extn_hfp_set_parameters(struct audio_device *adev,
struct str_parms *parms);
+int audio_extn_hfp_set_mic_mute(struct audio_device *adev, bool state);
+
#endif
#ifndef SOUND_TRIGGER_ENABLED
@@ -105,6 +109,9 @@
bool benable);
#endif
+void audio_extn_utils_send_default_app_type_cfg(void *platform, struct mixer *mixer);
+int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
+ struct audio_usecase *usecase);
#ifndef HWDEP_CAL_ENABLED
#define audio_extn_hwdep_cal_send(snd_card, acdb_handle) (0)
#else
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index ebc0fd2..07d5469 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -27,6 +27,7 @@
#include "platform_api.h"
#include <stdlib.h>
#include <cutils/str_parms.h>
+#include "audio_extn/tfa_98xx.h"
#define AUDIO_PARAMETER_HFP_ENABLE "hfp_enable"
#define AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
@@ -34,6 +35,10 @@
#define AUDIO_PARAMETER_HFP_VOL_MIXER_CTL "hfp_vol_mixer_ctl"
#define AUDIO_PARAMATER_HFP_VALUE_MAX 128
+#define AUDIO_PARAMETER_KEY_HFP_MIC_VOLUME "hfp_mic_volume"
+#define PLAYBACK_VOLUME_MAX 0x2000
+#define CAPTURE_VOLUME_DEFAULT (15.0)
+
static int32_t start_hfp(struct audio_device *adev,
struct str_parms *parms);
@@ -45,8 +50,10 @@
struct pcm *hfp_pcm_rx;
struct pcm *hfp_pcm_tx;
float hfp_volume;
+ float mic_volume;
char hfp_vol_mixer_ctl[AUDIO_PARAMATER_HFP_VALUE_MAX];
bool is_hfp_running;
+ bool mic_mute;
audio_usecase_t ucid;
};
@@ -56,8 +63,10 @@
.hfp_pcm_rx = NULL,
.hfp_pcm_tx = NULL,
.hfp_volume = 0,
+ .mic_volume = CAPTURE_VOLUME_DEFAULT,
.hfp_vol_mixer_ctl = {0, },
.is_hfp_running = 0,
+ .mic_mute = 0,
.ucid = USECASE_AUDIO_HFP_SCO,
};
static struct pcm_config pcm_config_hfp = {
@@ -79,6 +88,9 @@
ALOGV("%s: entry", __func__);
ALOGD("%s: (%f)\n", __func__, value);
+ hfpmod.hfp_volume = value;
+ audio_extn_tfa_98xx_set_voice_vol(value);
+
if (value < 0.0) {
ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
value = 0.0;
@@ -87,7 +99,6 @@
ALOGW("%s: Volume brought with in range (%f)\n", __func__, value);
}
vol = lrint((value * 0x2000) + 0.5);
- hfpmod.hfp_volume = value;
if (!hfpmod.is_hfp_running) {
ALOGV("%s: HFP not active, ignoring set_hfp_volume call", __func__);
@@ -119,6 +130,115 @@
return ret;
}
+
+/*Set mic volume to value.
+*
+* This interface is used for mic volume control, set mic volume as value(range 0 ~ 15).
+*/
+static int hfp_set_mic_volume(struct audio_device *adev, float value)
+{
+ int volume, ret = 0;
+ char mixer_ctl_name[128];
+ struct mixer_ctl *ctl;
+ int pcm_device_id = HFP_ASM_RX_TX;
+
+ if (!hfpmod.is_hfp_running) {
+ ALOGE("%s: HFP not active, ignoring set_hfp_mic_volume call", __func__);
+ return -EIO;
+ }
+
+ if (value < 0.0) {
+ ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
+ value = 0.0;
+ } else if (value > CAPTURE_VOLUME_DEFAULT) {
+ value = CAPTURE_VOLUME_DEFAULT;
+ ALOGW("%s: Volume brought within range (%f)\n", __func__, value);
+ }
+
+ value = value / CAPTURE_VOLUME_DEFAULT;
+ memset(mixer_ctl_name, 0, sizeof(mixer_ctl_name));
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Playback %d Volume", pcm_device_id);
+ 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 = (int)(value * PLAYBACK_VOLUME_MAX);
+
+ ALOGD("%s: Setting volume to %d (%s)\n", __func__, volume, mixer_ctl_name);
+ if (mixer_ctl_set_value(ctl, 0, volume) < 0) {
+ ALOGE("%s: Couldn't set HFP Volume: [%d]", __func__, volume);
+ return -EINVAL;
+ }
+
+ return ret;
+}
+
+static float hfp_get_mic_volume(struct audio_device *adev)
+{
+ int volume, ret = 0;
+ char mixer_ctl_name[128];
+ struct mixer_ctl *ctl;
+ int pcm_device_id = HFP_ASM_RX_TX;
+ float value = 0.0;
+
+ if (!hfpmod.is_hfp_running) {
+ ALOGE("%s: HFP not active, ignoring set_hfp_mic_volume call", __func__);
+ return -EIO;
+ }
+
+ memset(mixer_ctl_name, 0, sizeof(mixer_ctl_name));
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Playback %d Volume", pcm_device_id);
+ 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 = mixer_ctl_get_value(ctl, 0);
+ if ( volume < 0) {
+ ALOGE("%s: Couldn't set HFP Volume: [%d]", __func__, volume);
+ return -EINVAL;
+ }
+ ALOGD("%s: getting mic volume %d \n", __func__, volume);
+
+ value = (volume / PLAYBACK_VOLUME_MAX) * CAPTURE_VOLUME_DEFAULT;
+ if (value < 0.0) {
+ ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
+ value = 0.0;
+ } else if (value > CAPTURE_VOLUME_DEFAULT) {
+ value = CAPTURE_VOLUME_DEFAULT;
+ ALOGW("%s: Volume brought within range (%f)\n", __func__, value);
+ }
+
+ return value;
+}
+
+/*Set mic mute state.
+*
+* This interface is used for mic mute state control
+*/
+int audio_extn_hfp_set_mic_mute(struct audio_device *adev, bool state)
+{
+ int rc = 0;
+
+ if (state == hfpmod.mic_mute)
+ return rc;
+
+ if (state == true) {
+ hfpmod.mic_volume = hfp_get_mic_volume(adev);
+ }
+ rc = hfp_set_mic_volume(adev, (state == true) ? 0.0 : hfpmod.mic_volume);
+ adev->voice.mic_mute = state;
+ hfpmod.mic_mute = state;
+ ALOGD("%s: Setting mute state %d, rc %d\n", __func__, state, rc);
+ return rc;
+}
+
static int32_t start_hfp(struct audio_device *adev,
struct str_parms *parms __unused)
{
@@ -140,6 +260,8 @@
list_add_tail(&adev->usecase_list, &uc_info->list);
+ audio_extn_tfa_98xx_set_mode_bt();
+
select_devices(adev, hfpmod.ucid);
pcm_dev_rx_id = platform_get_pcm_device_id(uc_info->id, PCM_PLAYBACK);
@@ -169,13 +291,16 @@
}
ALOGD("%s: Opening PCM capture device card_id(%d) device_id(%d)",
__func__, adev->snd_card, pcm_dev_tx_id);
- hfpmod.hfp_pcm_rx = pcm_open(adev->snd_card,
- pcm_dev_rx_id,
- PCM_OUT, &pcm_config_hfp);
- if (hfpmod.hfp_pcm_rx && !pcm_is_ready(hfpmod.hfp_pcm_rx)) {
- ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_pcm_rx));
- ret = -EIO;
- goto exit;
+
+ if (audio_extn_tfa_98xx_is_supported() == false) {
+ hfpmod.hfp_pcm_rx = pcm_open(adev->snd_card,
+ pcm_dev_rx_id,
+ PCM_OUT, &pcm_config_hfp);
+ if (hfpmod.hfp_pcm_rx && !pcm_is_ready(hfpmod.hfp_pcm_rx)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_pcm_rx));
+ ret = -EIO;
+ goto exit;
+ }
}
hfpmod.hfp_sco_tx = pcm_open(adev->snd_card,
pcm_dev_asm_tx_id,
@@ -187,22 +312,33 @@
}
ALOGV("%s: Opening PCM capture device card_id(%d) device_id(%d)",
__func__, adev->snd_card, pcm_dev_tx_id);
- hfpmod.hfp_pcm_tx = pcm_open(adev->snd_card,
- pcm_dev_tx_id,
- PCM_IN, &pcm_config_hfp);
- if (hfpmod.hfp_pcm_tx && !pcm_is_ready(hfpmod.hfp_pcm_tx)) {
- ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_pcm_tx));
- ret = -EIO;
- goto exit;
+
+ if (audio_extn_tfa_98xx_is_supported() == false) {
+ hfpmod.hfp_pcm_tx = pcm_open(adev->snd_card,
+ pcm_dev_tx_id,
+ PCM_IN, &pcm_config_hfp);
+ if (hfpmod.hfp_pcm_tx && !pcm_is_ready(hfpmod.hfp_pcm_tx)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_pcm_tx));
+ ret = -EIO;
+ goto exit;
+ }
}
pcm_start(hfpmod.hfp_sco_rx);
pcm_start(hfpmod.hfp_sco_tx);
- pcm_start(hfpmod.hfp_pcm_rx);
- pcm_start(hfpmod.hfp_pcm_tx);
+ if (audio_extn_tfa_98xx_is_supported() == false) {
+ pcm_start(hfpmod.hfp_pcm_rx);
+ pcm_start(hfpmod.hfp_pcm_tx);
+ }
+
+ audio_extn_tfa_98xx_enable_speaker();
hfpmod.is_hfp_running = true;
hfp_set_volume(adev, hfpmod.hfp_volume);
+ /* Set mic volume by mute status, we don't provide set mic volume in phone app, only
+ provide mute and unmute. */
+ audio_extn_hfp_set_mic_mute(adev, adev->mic_muted);
+
ALOGD("%s: exit: status(%d)", __func__, ret);
return 0;
@@ -348,6 +484,20 @@
ALOGD("%s: set_hfp_volume usecase, Vol: [%f]", __func__, vol);
hfp_set_volume(adev, vol);
}
+
+ memset(value, 0, sizeof(value));
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_HFP_MIC_VOLUME,
+ value, sizeof(value));
+ if (ret >= 0) {
+ if (sscanf(value, "%f", &vol) != 1){
+ ALOGE("%s: error in retrieving hfp mic volume", __func__);
+ ret = -EIO;
+ goto exit;
+ }
+ ALOGD("%s: set_hfp_mic_volume usecase, Vol: [%f]", __func__, vol);
+ hfp_set_mic_volume(adev, vol);
+ }
+
exit:
ALOGV("%s Exit",__func__);
}
diff --git a/hal/audio_extn/tfa_98xx.c b/hal/audio_extn/tfa_98xx.c
new file mode 100644
index 0000000..41523b5
--- /dev/null
+++ b/hal/audio_extn/tfa_98xx.c
@@ -0,0 +1,545 @@
+/*
+ * Copyright (C) 2013-2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "tfa_98xx"
+/*#define LOG_NDEBUG 0*/
+#include <cutils/log.h>
+
+#include <stdlib.h>
+#include <audio_hw.h>
+#include <dlfcn.h>
+#include "audio_extn.h"
+#include <platform.h>
+#include <math.h>
+
+#define LIB_SPEAKER_BUNDLE "/system/lib/libexTfa98xx.so"
+
+
+enum exTfa98xx_Audio_Mode
+{
+ Audio_Mode_None = -1,
+ Audio_Mode_Music_Normal,
+ Audio_Mode_Hfp_Client,
+ Audio_Mode_Voice,
+ Audio_Mode_Hs_Hfp,
+ Audio_Mode_Max
+};
+typedef enum exTfa98xx_Audio_Mode exTfa98xx_audio_mode_t;
+
+enum exTfa98xx_Func_Mode
+{
+ Func_Mode_None = -1,
+ Func_Mode_Speaker,
+ Func_Mode_BT
+};
+typedef enum exTfa98xx_Func_Mode exTfa98xx_func_mode_t;
+
+#define I2S_CLOCK_ENABLE 1
+#define I2S_CLOCK_DISABLE 0
+#define HFP_MAX_VOLUME (15.000000)
+#define TFA_98XX_HFP_VSETPS (5.0)
+
+exTfa98xx_audio_mode_t current_audio_mode = Audio_Mode_None;
+
+typedef int (*set_speaker_on_t)(exTfa98xx_audio_mode_t);
+typedef int (*set_speaker_off_t)(void);
+typedef int (*set_speaker_calibration_t)(int);
+typedef void (*set_speaker_volume_step_t)(int, int);
+
+
+struct speaker_data {
+ struct audio_device *adev;
+ void *speaker_bundle;
+ set_speaker_on_t set_speaker_on;
+ set_speaker_off_t set_speaker_off;
+ set_speaker_calibration_t set_speaker_calibration;
+ set_speaker_volume_step_t set_speaker_volume_step;
+ int ref_cnt[Audio_Mode_Max];
+ int route_cnt[Audio_Mode_Max];
+ bool update_ref_cnt;
+};
+
+struct speaker_data *tfa98xx_speaker_data = NULL;
+
+static struct speaker_data* open_speaker_bundle()
+{
+ struct speaker_data *sd = calloc(1, sizeof(struct speaker_data));
+
+ sd->speaker_bundle = dlopen(LIB_SPEAKER_BUNDLE, RTLD_NOW);
+ if (sd->speaker_bundle == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, LIB_SPEAKER_BUNDLE);
+ goto error;
+ } else {
+ ALOGV("%s: DLOPEN successful for %s", __func__, LIB_SPEAKER_BUNDLE);
+
+ sd->set_speaker_on = (set_speaker_on_t)dlsym(sd->speaker_bundle,
+ "exTfa98xx_speakeron");
+ if (sd->set_speaker_on == NULL) {
+ ALOGE("%s: dlsym error %s for exTfa98xx_speakeron", __func__,
+ dlerror());
+ goto error;
+ }
+ sd->set_speaker_off = (set_speaker_off_t)dlsym(sd->speaker_bundle,
+ "exTfa98xx_speakeroff");
+ if (sd->set_speaker_off == NULL) {
+ ALOGE("%s: dlsym error %s for exTfa98xx_speakeroff", __func__,
+ dlerror());
+ goto error;
+ }
+ sd->set_speaker_volume_step = (set_speaker_volume_step_t)dlsym(sd->speaker_bundle,
+ "exTfa98xx_setvolumestep");
+ if (sd->set_speaker_volume_step == NULL) {
+ ALOGE("%s: dlsym error %s for exTfa98xx_setvolumestep",
+ __func__, dlerror());
+ goto error;
+ }
+ sd->set_speaker_calibration = (set_speaker_calibration_t)dlsym(sd->speaker_bundle,
+ "exTfa98xx_calibration");
+ if (sd->set_speaker_calibration == NULL) {
+ ALOGE("%s: dlsym error %s for exTfa98xx_calibration",
+ __func__, dlerror());
+ goto error;
+ }
+ }
+ return sd;
+
+error:
+ free(sd);
+ return 0;
+}
+
+static void close_speaker_bundle(struct speaker_data *sd)
+{
+ if (sd != NULL) {
+ dlclose(sd->speaker_bundle);
+ free(sd);
+ sd = NULL;
+ }
+}
+
+static int adev_i2s_clock_operation(int enable, struct audio_device *adev, char *paths)
+{
+ int ret = -1;
+
+ ALOGD("%s: mixer paths is: %s, enable: %d\n", __func__, paths, enable);
+ if(I2S_CLOCK_ENABLE == enable) {
+ ret = audio_route_apply_and_update_path(adev->audio_route, paths);
+ if(ret) {
+ ALOGE("%s: audio_route_apply_and_update_path return %d\n", __func__, ret);
+ return ret;
+ }
+ } else {
+ ret = audio_route_reset_and_update_path(adev->audio_route, paths);
+ if(ret) {
+ ALOGE("%s: audio_route_reset_and_update_path return %d\n", __func__, ret);
+ return ret;
+ }
+ }
+ return 0;
+}
+
+static int tfa_98xx_set_audio_mode(int enable, struct audio_device *adev, exTfa98xx_audio_mode_t audio_mode)
+{
+ char paths[32] = "init_smart_pa";
+
+ switch(audio_mode) {
+ case Audio_Mode_Music_Normal:
+ strcat(paths, " music");
+ break;
+ case Audio_Mode_Voice:
+ case Audio_Mode_Hfp_Client:
+ case Audio_Mode_Hs_Hfp:
+ strcat(paths, " voice");
+ break;
+ default:
+ ALOGE("%s: function %d not support!\n",__func__, audio_mode);
+ return -EINVAL;
+ }
+
+ ALOGV("%s: mixer paths is: %s, enable: %d\n", __func__, paths, enable);
+ adev_i2s_clock_operation(enable, adev, paths);
+ return 0;
+
+}
+
+static exTfa98xx_audio_mode_t tfa_98xx_get_audio_mode(struct speaker_data *data)
+{
+ exTfa98xx_audio_mode_t tfa_98xx_audio_mode = Audio_Mode_None;
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ audio_mode_t mode = data->adev->mode;
+ int i = 0;
+
+ ALOGV("%s: enter\n", __func__);
+
+ for (i = 0; i < Audio_Mode_Max; i++)
+ data->route_cnt[i] = 0;
+
+ list_for_each(node, &data->adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->devices & AUDIO_DEVICE_OUT_ALL_SCO) {
+ if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+ tfa_98xx_audio_mode = Audio_Mode_Hs_Hfp;
+ data->route_cnt[tfa_98xx_audio_mode]++;
+ ALOGV("%s: audio_mode hs_hfp\n", __func__);
+ }
+ } else if (usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) {
+ if ((mode == AUDIO_MODE_IN_CALL) || audio_extn_hfp_is_active(data->adev)) {
+ if (audio_extn_hfp_is_active(data->adev)) {
+ if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+ tfa_98xx_audio_mode = Audio_Mode_Hfp_Client;
+ data->route_cnt[tfa_98xx_audio_mode]++;
+ ALOGV("%s: audio_mode hfp client\n", __func__);
+ }
+ } else {
+ if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+ tfa_98xx_audio_mode = Audio_Mode_Voice;
+ data->route_cnt[tfa_98xx_audio_mode]++;
+ ALOGV("%s: audio_mode voice\n", __func__);
+ }
+ }
+ } else {
+ if (data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+ tfa_98xx_audio_mode = Audio_Mode_Music_Normal;
+ data->route_cnt[tfa_98xx_audio_mode]++;
+ ALOGV("%s: tfa_98xx_audio_mode music\n", __func__);
+ }
+ }
+ } else {
+ ALOGE("%s: no device match \n", __func__);
+ }
+ }
+ ALOGV("%s: tfa_98xx_audio_mode %d exit\n", __func__, tfa_98xx_audio_mode);
+
+ return tfa_98xx_audio_mode;
+}
+
+static int tfa_98xx_set_func_mode(int enable, struct audio_device *adev, exTfa98xx_func_mode_t func_mode)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ char paths[32] = "init_smart_pa";
+
+ if (data) {
+ switch(func_mode) {
+ case Func_Mode_Speaker:
+ strcat(paths, " func_speaker");
+ break;
+ case Func_Mode_BT:
+ strcat(paths, " func_bt");
+ break;
+ default:
+ ALOGE("%s: function %d not support!\n",__func__, func_mode);
+ return -EINVAL;
+ }
+
+ ALOGV("%s: mixer paths is: %s, enable: %d\n", __func__, paths, enable);
+ adev_i2s_clock_operation(enable, adev, paths);
+ }
+ return 0;
+}
+
+static exTfa98xx_func_mode_t tfa_98xx_get_func_mode(exTfa98xx_audio_mode_t audio_mode)
+{
+ exTfa98xx_func_mode_t func_mode = Func_Mode_None;
+
+ switch(audio_mode) {
+ case Audio_Mode_Music_Normal:
+ case Audio_Mode_Voice:
+ ALOGV("%s: tfa_98xx_func_mode speaker \n", __func__);
+ func_mode = Func_Mode_Speaker;
+ break;
+ case Audio_Mode_Hfp_Client:
+ case Audio_Mode_Hs_Hfp:
+ ALOGV("%s: tfa_98xx_func_mode bt \n", __func__);
+ func_mode = Func_Mode_BT;
+ break;
+ default:
+ break;
+ }
+ return func_mode;
+}
+
+static void tfa_98xx_disable_speaker(void)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ int ret = 0;
+
+ ret = data->set_speaker_off();
+ if (ret) {
+ ALOGE("%s: exTfa98xx_speakeroff failed result = %d\n", __func__, ret);
+ goto on_error;
+ }
+
+ ret = tfa_98xx_set_audio_mode(I2S_CLOCK_DISABLE, data->adev, current_audio_mode);
+ if (ret) {
+ ALOGE("%s: tfa_98xx_set_audio_mode disable failed return %d\n", __func__, ret);
+ goto on_error;
+ }
+ current_audio_mode = Audio_Mode_None;
+on_error:
+ return;
+
+}
+
+
+void audio_extn_tfa_98xx_disable_speaker(snd_device_t snd_device)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ int i = 0;
+ exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_None;
+
+ ALOGV("%s: enter\n", __func__);
+
+ if (data) {
+ if ((current_audio_mode == Audio_Mode_None) || (snd_device > SND_DEVICE_OUT_END))
+ goto on_exit;
+
+ switch(snd_device) {
+ case SND_DEVICE_OUT_SPEAKER:
+ new_audio_mode = Audio_Mode_Music_Normal;
+ break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER:
+ new_audio_mode = Audio_Mode_Voice;
+ break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_HFP:
+ new_audio_mode = Audio_Mode_Hfp_Client;
+ break;
+ case SND_DEVICE_OUT_BT_SCO:
+ new_audio_mode = Audio_Mode_Hs_Hfp;
+ break;
+ default:
+ break;
+ }
+
+ if ((new_audio_mode == Audio_Mode_None) || (data->ref_cnt[new_audio_mode] <= 0)) {
+ ALOGE("%s: device ref cnt is already 0", __func__);
+ goto on_exit;
+ }
+
+ data->ref_cnt[new_audio_mode]--;
+
+ for (i = 0; i < Audio_Mode_Max; i++) {
+ if (data->ref_cnt[i] > 0) {
+ ALOGD("%s: exTfa98xx_speaker still in use\n", __func__);
+ goto on_exit;
+ }
+ }
+
+ if (data->adev->enable_hfp)
+ data->set_speaker_volume_step(0, 0);
+
+ tfa_98xx_disable_speaker();
+ }
+
+ ALOGV("%s: exit\n", __func__);
+on_exit:
+ return;
+}
+
+int audio_extn_tfa_98xx_enable_speaker(void)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_Music_Normal;
+ int ret = 0;
+ int i = 0;
+
+ ALOGV("%s: enter\n", __func__);
+
+ if (data) {
+
+ new_audio_mode = tfa_98xx_get_audio_mode(data);
+ if ((new_audio_mode != Audio_Mode_None) && (data->ref_cnt[new_audio_mode] >= 1)) {
+ ALOGD("%s, mode %d already active!", __func__, new_audio_mode);
+ data->ref_cnt[new_audio_mode]++;
+ goto on_exit;
+ }
+
+ ret = tfa_98xx_set_audio_mode(I2S_CLOCK_ENABLE, data->adev, new_audio_mode);
+ if (ret) {
+ ALOGE("%s: tfa_98xx_set_audio_mode enable failed return %d\n", __func__, ret);
+ goto on_exit;
+ }
+
+ ret = data->set_speaker_on(new_audio_mode);
+ if (ret) {
+ ALOGE("%s: exTfa98xx_speakeron failed result = %d\n", __func__, ret);
+ goto on_exit;
+ }
+
+ current_audio_mode = new_audio_mode;
+ for (i = 0; i < Audio_Mode_Max; i++) {
+ data->ref_cnt[i] = data->route_cnt[i];
+ }
+ data->update_ref_cnt = false;
+ }
+
+ ALOGV("%s: exit\n", __func__);
+
+on_exit:
+ return ret;
+
+}
+
+void audio_extn_tfa_98xx_set_mode(void)
+{
+ int ret = 0;
+ struct speaker_data *data = tfa98xx_speaker_data;
+ exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_None;
+ exTfa98xx_func_mode_t new_func_mode = Func_Mode_None;
+
+ ALOGV("%s: enter\n", __func__);
+
+ if (data) {
+ new_audio_mode = tfa_98xx_get_audio_mode(data);
+
+ new_func_mode = tfa_98xx_get_func_mode(new_audio_mode);
+ if (new_func_mode == Func_Mode_None)
+ return;
+
+ ret = tfa_98xx_set_func_mode(I2S_CLOCK_ENABLE, data->adev, new_func_mode);
+ if (ret) {
+ ALOGE("%s: tfa_98xx_set_func_mode enable return %d\n", __func__, ret);
+ }
+ data->update_ref_cnt = true;
+ }
+
+ ALOGV("%s: exit\n", __func__);
+}
+
+void audio_extn_tfa_98xx_set_mode_bt(void)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ int ret = 0;
+
+ if (data) {
+ ret = tfa_98xx_set_func_mode(I2S_CLOCK_ENABLE, data->adev, Func_Mode_BT);
+ if (ret) {
+ ALOGE("%s: tfa_98xx_set_func_mode enable return %d\n", __func__, ret);
+ }
+ }
+}
+
+void audio_extn_tfa_98xx_update(void)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_Music_Normal;
+
+ ALOGD("%s: enter\n", __func__);
+
+ if (data) {
+
+ new_audio_mode = tfa_98xx_get_audio_mode(data);
+ if (new_audio_mode <= current_audio_mode) {
+ ALOGE("%s: audio_extn_tfa_98xx_update same mode\n", __func__);
+ if (data->update_ref_cnt == true) {
+ data->ref_cnt[new_audio_mode]++;
+ data->update_ref_cnt = false;
+ }
+ goto on_error;
+ }
+
+ if (current_audio_mode != Audio_Mode_None) {
+ tfa_98xx_disable_speaker();
+ }
+
+ audio_extn_tfa_98xx_enable_speaker();
+
+ }
+
+ ALOGV("%s: exit\n", __func__);
+on_error:
+ return;
+
+}
+
+void audio_extn_tfa_98xx_set_voice_vol(float vol)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ int vsteps = 0;
+
+ if (data) {
+ if (data->adev->enable_hfp) {
+ if (vol < 0.0) {
+ vol = 0.0;
+ } else {
+ vol = ((vol > HFP_MAX_VOLUME) ? 1.0 : (vol / HFP_MAX_VOLUME));
+ }
+ vsteps = (int)floorf((1.0 - vol) * TFA_98XX_HFP_VSETPS);
+ } else {
+ return;
+ }
+ ALOGD("%s: vsteps %d\n", __func__, vsteps);
+ data->set_speaker_volume_step(vsteps, vsteps);
+ }
+}
+
+bool audio_extn_tfa_98xx_is_supported(void)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+ if (data)
+ return true;
+ else
+ return false;
+}
+
+int audio_extn_tfa_98xx_init(struct audio_device *adev)
+{
+ int ret = 0;
+ struct speaker_data *data = open_speaker_bundle();
+
+ ALOGV("%s: enter\n", __func__);
+
+ if (data) {
+ ret = tfa_98xx_set_audio_mode(I2S_CLOCK_ENABLE, adev, Audio_Mode_Music_Normal);
+ if (ret) {
+ ALOGE("%s: tfa_98xx_set_audio_mode enable return %d\n", __func__, ret);
+ goto err_init;
+ }
+
+ ret = data->set_speaker_calibration(0);
+ if (ret) {
+ ALOGE("%s: exTfa98xx_calibration return %d\n", __func__, ret);
+ }
+
+ ret = tfa_98xx_set_audio_mode(I2S_CLOCK_DISABLE, adev, Audio_Mode_Music_Normal);
+ if (ret) {
+ ALOGE("%s: tfa_98xx_set_audio_mode disable return %d\n", __func__, ret);
+ goto err_init;
+ }
+
+ data->adev = adev;
+ tfa98xx_speaker_data = data;
+ ALOGV("%s: exit\n", __func__);
+ return 0;
+
+ }
+
+err_init:
+ close_speaker_bundle(data);
+ return -EINVAL;
+}
+
+void audio_extn_tfa_98xx_deinit(void)
+{
+ struct speaker_data *data = tfa98xx_speaker_data;
+
+ if (data) {
+ data->set_speaker_off();
+ close_speaker_bundle(data);
+ tfa98xx_speaker_data = NULL;
+ }
+}
+
diff --git a/hal/audio_extn/tfa_98xx.h b/hal/audio_extn/tfa_98xx.h
new file mode 100644
index 0000000..444429a
--- /dev/null
+++ b/hal/audio_extn/tfa_98xx.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2013-2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TFA_98XX_H
+#define TFA_98XX_H
+
+#ifdef SMART_PA_TFA_98XX_SUPPORTED
+int audio_extn_tfa_98xx_enable_speaker(void);
+void audio_extn_tfa_98xx_disable_speaker(snd_device_t snd_device);
+void audio_extn_tfa_98xx_set_mode();
+void audio_extn_tfa_98xx_set_mode_bt(void);
+void audio_extn_tfa_98xx_update(void);
+void audio_extn_tfa_98xx_set_voice_vol(float vol);
+int audio_extn_tfa_98xx_init(struct audio_device *adev);
+void audio_extn_tfa_98xx_deinit(void);
+bool audio_extn_tfa_98xx_is_supported(void);
+#else
+#define audio_extn_tfa_98xx_enable_speaker(void) (0)
+#define audio_extn_tfa_98xx_disable_speaker(snd_device) (0)
+#define audio_extn_tfa_98xx_set_mode() (0)
+#define audio_extn_tfa_98xx_set_mode_bt() (0)
+#define audio_extn_tfa_98xx_update(void) (0)
+#define audio_extn_tfa_98xx_set_voice_vol(vol) (0)
+#define audio_extn_tfa_98xx_init(adev) (0)
+#define audio_extn_tfa_98xx_deinit(void) (0)
+#define audio_extn_tfa_98xx_is_supported(void) (false)
+#endif
+
+#endif /* TFA_98XX_H */
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
new file mode 100644
index 0000000..d2bb42c
--- /dev/null
+++ b/hal/audio_extn/utils.c
@@ -0,0 +1,179 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_utils"
+//#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <cutils/properties.h>
+#include <cutils/config_utils.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+#include <cutils/misc.h>
+
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include "audio_extn.h"
+
+#define MAX_LENGTH_MIXER_CONTROL_IN_INT 128
+
+static int set_mixer_ctrl(struct audio_device *adev,
+ int pcm_device_id, int app_type,
+ int acdb_dev_id, int sample_rate, int stream_type)
+{
+
+ char mixer_ctl_name[MAX_LENGTH_MIXER_CONTROL_IN_INT];
+ struct mixer_ctl *ctl;
+ int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc = 0;
+
+ if (stream_type == PCM_PLAYBACK) {
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Audio Stream %d App Type Cfg", pcm_device_id);
+ } else if (stream_type == PCM_CAPTURE) {
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Audio Stream Capture %d App Type Cfg", pcm_device_id);
+ }
+
+ 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);
+ rc = -EINVAL;
+ goto exit;
+ }
+ app_type_cfg[len++] = app_type;
+ app_type_cfg[len++] = acdb_dev_id;
+ app_type_cfg[len++] = sample_rate;
+ ALOGV("%s: stream type %d app_type %d, acdb_dev_id %d sample rate %d",
+ __func__, stream_type, app_type, acdb_dev_id, sample_rate);
+ mixer_ctl_set_array(ctl, app_type_cfg, len);
+
+exit:
+ return rc;
+}
+
+void audio_extn_utils_send_default_app_type_cfg(void *platform, struct mixer *mixer)
+{
+ int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT] = {-1};
+ int length = 0, app_type = 0,rc = 0;
+ struct mixer_ctl *ctl = NULL;
+ const char *mixer_ctl_name = "App Type Config";
+
+ ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",__func__, mixer_ctl_name);
+ return;
+ }
+ rc = platform_get_default_app_type_v2(platform, PCM_PLAYBACK, &app_type);
+ if (rc == 0) {
+ app_type_cfg[length++] = 1;
+ app_type_cfg[length++] = app_type;
+ app_type_cfg[length++] = 48000;
+ app_type_cfg[length++] = 16;
+ mixer_ctl_set_array(ctl, app_type_cfg, length);
+ }
+ return;
+}
+
+int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
+ struct audio_usecase *usecase)
+{
+ struct mixer_ctl *ctl;
+ int pcm_device_id, acdb_dev_id = 0, snd_device = usecase->out_snd_device;
+ int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ int app_type = 0, rc = 0;
+
+ ALOGV("%s", __func__);
+
+ if (usecase->type != PCM_HFP_CALL) {
+ ALOGV("%s: not a playback or HFP path, no need to cfg app type", __func__);
+ rc = 0;
+ goto exit_send_app_type_cfg;
+ }
+ if ((usecase->id != USECASE_AUDIO_HFP_SCO) &&
+ (usecase->id != USECASE_AUDIO_HFP_SCO_WB)) {
+ ALOGV("%s: a playback path where app type cfg is not required", __func__);
+ rc = 0;
+ goto exit_send_app_type_cfg;
+ }
+
+ snd_device = usecase->out_snd_device;
+ pcm_device_id = platform_get_pcm_device_id(usecase->id, PCM_PLAYBACK);
+
+ snd_device = (snd_device == SND_DEVICE_OUT_SPEAKER) ?
+ audio_extn_get_spkr_prot_snd_device(snd_device) : snd_device;
+ acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+ if (acdb_dev_id < 0) {
+ ALOGE("%s: Couldn't get the acdb dev id", __func__);
+ rc = -EINVAL;
+ goto exit_send_app_type_cfg;
+ }
+
+ if (usecase->type == PCM_HFP_CALL) {
+
+ /* config HFP session:1 playback path */
+ rc = platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK, &app_type);
+ if (rc < 0)
+ goto exit_send_app_type_cfg;
+
+ sample_rate= CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ rc = set_mixer_ctrl(adev, pcm_device_id, app_type,
+ acdb_dev_id, sample_rate, PCM_PLAYBACK);
+ if (rc < 0)
+ goto exit_send_app_type_cfg;
+ /* config HFP session:1 capture path */
+ rc = platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE, &app_type);
+
+ if (rc == 0) {
+ rc = set_mixer_ctrl(adev, pcm_device_id, app_type,
+ acdb_dev_id, sample_rate, PCM_CAPTURE);
+ if (rc < 0)
+ goto exit_send_app_type_cfg;
+ }
+ /* config HFP session:2 capture path */
+ pcm_device_id = HFP_ASM_RX_TX;
+ snd_device = usecase->in_snd_device;
+ acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+ if (acdb_dev_id <= 0) {
+ ALOGE("%s: Couldn't get the acdb dev id", __func__);
+ rc = -EINVAL;
+ goto exit_send_app_type_cfg;
+ }
+ rc = platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE, &app_type);
+ if (rc == 0) {
+ rc = set_mixer_ctrl(adev, pcm_device_id, app_type,
+ acdb_dev_id, sample_rate, PCM_CAPTURE);
+ if (rc < 0)
+ goto exit_send_app_type_cfg;
+ }
+
+ /* config HFP session:2 playback path */
+ rc = platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK, &app_type);
+ if (rc == 0) {
+ rc = set_mixer_ctrl(adev, pcm_device_id, app_type,
+ acdb_dev_id, sample_rate, PCM_PLAYBACK);
+ if (rc < 0)
+ goto exit_send_app_type_cfg;
+ }
+ }
+
+ rc = 0;
+exit_send_app_type_cfg:
+ return rc;
+}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 6770e03..7b10f56 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -53,6 +53,7 @@
#include "voice_extn.h"
#include "sound/compress_params.h"
+#include "audio_extn/tfa_98xx.h"
/* COMPRESS_OFFLOAD_FRAGMENT_SIZE must be more than 8KB and a multiple of 32KB if more than 32KB.
* COMPRESS_OFFLOAD_FRAGMENT_SIZE * COMPRESS_OFFLOAD_NUM_FRAGMENTS must be less than 8MB. */
@@ -531,6 +532,18 @@
return id;
}
+static int audio_ssr_status(struct audio_device *adev)
+{
+ int ret = 0;
+ struct mixer_ctl *ctl;
+ const char *mixer_ctl_name = "Audio SSR Status";
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ ret = mixer_ctl_get_value(ctl, 0);
+ ALOGD("%s: value: %d", __func__, ret);
+ return ret;
+}
+
int enable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase)
{
@@ -547,6 +560,7 @@
else
snd_device = usecase->out_snd_device;
+ audio_extn_utils_send_app_type_cfg(adev, usecase);
strcpy(mixer_path, use_case_table[usecase->id]);
platform_add_backend_name(adev->platform, mixer_path, snd_device);
ALOGD("%s: usecase(%d) apply and update mixer path: %s", __func__, usecase->id, mixer_path);
@@ -668,6 +682,8 @@
ALOGE("%s: device ref cnt is already 0", __func__);
return -EINVAL;
}
+ audio_extn_tfa_98xx_disable_speaker(snd_device);
+
adev->snd_dev_ref_cnt[snd_device]--;
if (adev->snd_dev_ref_cnt[snd_device] == 0) {
audio_extn_dsm_feedback_enable(adev, snd_device, false);
@@ -1186,6 +1202,8 @@
usecase->in_snd_device = in_snd_device;
usecase->out_snd_device = out_snd_device;
+ audio_extn_tfa_98xx_set_mode();
+
enable_audio_route(adev, usecase);
/* Applicable only on the targets that has external modem.
@@ -1243,6 +1261,9 @@
ALOGV("%s: enter: usecase(%d)", __func__, in->usecase);
+ if (audio_extn_tfa_98xx_is_supported() && !audio_ssr_status(adev))
+ return -EIO;
+
if (in->card_status == CARD_STATUS_OFFLINE ||
adev->card_status == CARD_STATUS_OFFLINE) {
ALOGW("in->card_status or adev->card_status offline, try again");
@@ -1735,6 +1756,8 @@
}
register_out_stream(out);
audio_extn_perf_lock_release();
+ audio_extn_tfa_98xx_enable_speaker();
+
ALOGV("%s: exit", __func__);
return ret;
error_open:
@@ -2039,6 +2062,7 @@
out->routing_change = true;
}
select_devices(adev, out->usecase);
+ audio_extn_tfa_98xx_update();
}
}
@@ -2197,14 +2221,36 @@
const void *buffer, size_t bytes)
{
struct stream_out *out = (struct stream_out *)stream;
+ struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
+ int64_t now;
+ int64_t elapsed_time_since_last_write = 0;
+ int64_t sleep_time;
- /* No Output device supported other than BT for playback.
- * Sleep for the amount of buffer duration
- */
+ clock_gettime(CLOCK_MONOTONIC, &t);
+ now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
+
lock_output_stream(out);
- usleep(bytes * 1000000 / audio_stream_out_frame_size(&out->stream.common) /
- out_get_sample_rate(&out->stream.common));
+ if (out->last_write_time_us)
+ elapsed_time_since_last_write = now - out->last_write_time_us;
+ sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) /
+ out_get_sample_rate(&stream->common) - elapsed_time_since_last_write;
+ if (sleep_time > 0) {
+ usleep(sleep_time);
+ } else {
+ // we don't sleep when we exit standby (this is typical for a real alsa buffer).
+ sleep_time = 0;
+ }
+ out->last_write_time_us = now + sleep_time;
pthread_mutex_unlock(&out->lock);
+ // last_write_time_us is an approximation of when the (simulated) alsa
+ // buffer is believed completely full. The usleep above waits for more space
+ // in the buffer, but by the end of the sleep the buffer is considered
+ // topped-off.
+ //
+ // On the subsequent out_write(), we measure the elapsed time spent in
+ // the mixer. This is subtracted from the sleep estimate based on frames,
+ // thereby accounting for drain in the alsa buffer during mixing.
+ // This is a crude approximation; we don't handle underruns precisely.
return bytes;
}
#endif
@@ -3279,7 +3325,11 @@
ALOGD("%s: state %d", __func__, (int)state);
pthread_mutex_lock(&adev->lock);
- ret = voice_set_mic_mute(adev, state);
+ if (audio_extn_tfa_98xx_is_supported() && adev->enable_hfp) {
+ ret = audio_extn_hfp_set_mic_mute(adev, state);
+ } else {
+ ret = voice_set_mic_mute(adev, state);
+ }
adev->mic_muted = state;
pthread_mutex_unlock(&adev->lock);
@@ -3321,6 +3371,9 @@
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
return -EINVAL;
+ if (audio_extn_tfa_98xx_is_supported() && (audio_extn_hfp_is_active(adev) || voice_is_in_call(adev)))
+ return -EINVAL;
+
in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
@@ -3619,6 +3672,8 @@
if (!adev)
return 0;
+ audio_extn_tfa_98xx_deinit();
+
audio_extn_snd_mon_unregister_listener(adev);
pthread_mutex_lock(&adev_init_lock);
@@ -3837,6 +3892,7 @@
}
}
+ audio_extn_utils_send_default_app_type_cfg(adev->platform, adev->mixer);
audio_device_ref_count++;
if (property_get("audio_hal.period_multiplier", value, NULL) > 0) {
@@ -3849,6 +3905,8 @@
ALOGV("new period_multiplier = %d", af_period_multiplier);
}
+ audio_extn_tfa_98xx_init(adev);
+
pthread_mutex_unlock(&adev_init_lock);
if (adev->adm_init)
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 4963316..2cf72e1 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -210,6 +210,8 @@
card_status_t card_status;
struct error_log error_log;
+
+ int64_t last_write_time_us;
};
struct stream_in {
@@ -240,7 +242,7 @@
card_status_t card_status;
};
-typedef enum {
+typedef enum usecase_type_t {
PCM_PLAYBACK,
PCM_CAPTURE,
VOICE_CALL,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 72d9591..4995bd8 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -31,6 +31,7 @@
#include "audio_extn.h"
#include "voice_extn.h"
#include "sound/msmcal-hwdep.h"
+#include "audio_extn/tfa_98xx.h"
#include <dirent.h>
#define MAX_MIXER_XML_PATH 100
#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
@@ -56,6 +57,10 @@
/* EDID format ID for LPCM audio */
#define EDID_FORMAT_LPCM 1
+/* fallback app type if the default app type from acdb loader fails */
+#define DEFAULT_APP_TYPE_RX_PATH 0x11130
+#define DEFAULT_APP_TYPE_TX_PATH 0x11132
+
/* Retry for delay in FW loading*/
#define RETRY_NUMBER 20
#define RETRY_US 500000
@@ -89,6 +94,11 @@
CAL_MODE_RTAC = 0x4
};
+enum {
+ BUFF_IDX_0 = 0,
+ BUFF_IDX_1 = 1,
+};
+
#define PLATFORM_CONFIG_KEY_OPERATOR_INFO "operator_info"
struct operator_info {
@@ -111,6 +121,7 @@
typedef void (*acdb_deallocate_t)();
typedef int (*acdb_init_v2_cvd_t)(const char *, char *, int);
typedef void (*acdb_send_audio_cal_t)(int, int);
+typedef void (*acdb_send_audio_cal_v3_t)(int, int, int , int, int);
typedef void (*acdb_send_voice_cal_t)(int, int);
typedef int (*acdb_reload_vocvoltable_t)(int);
typedef int (*acdb_loader_get_calibration_t)(char *attr, int size, void *data);
@@ -131,6 +142,7 @@
acdb_init_v2_cvd_t acdb_init;
acdb_deallocate_t acdb_deallocate;
acdb_send_audio_cal_t acdb_send_audio_cal;
+ acdb_send_audio_cal_v3_t acdb_send_audio_cal_v3;
acdb_send_voice_cal_t acdb_send_voice_cal;
acdb_reload_vocvoltable_t acdb_reload_vocvoltable;
void *hw_info;
@@ -293,7 +305,7 @@
[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 14,
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
- [SND_DEVICE_OUT_VOICE_SPEAKER_HFP] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_HFP] = 140,
[SND_DEVICE_IN_HANDSET_MIC] = 4,
[SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = 4,
@@ -323,7 +335,7 @@
[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = 123,
[SND_DEVICE_IN_CAMCORDER_MIC] = 4,
[SND_DEVICE_IN_VOICE_DMIC] = 41,
- [SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP] = 11,
+ [SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP] = 141,
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = 43,
[SND_DEVICE_IN_VOICE_SPEAKER_QMIC] = 19,
[SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC] = 16,
@@ -540,7 +552,7 @@
char mccmnc[PROPERTY_VALUE_MAX];
char *ret = NULL;
- property_get("gsm.sim.operator.numeric",mccmnc,"0");
+ property_get("gsm.sim.operator.numeric",mccmnc,"00000");
list_for_each(node, &operator_info_list) {
info_item = node_to_item(node, struct operator_info, list);
@@ -831,24 +843,39 @@
}
list_init(&operator_info_list);
+ bool card_verifed[MAX_SND_CARD] = {0};
+ const int retry_limit = property_get_int32("audio.snd_card.open.retries", RETRY_NUMBER);
- while (snd_card_num < MAX_SND_CARD) {
- adev->mixer = mixer_open(snd_card_num);
+ for (;;) {
+ if (snd_card_num >= MAX_SND_CARD) {
+ if (retry_num++ >= retry_limit) {
+ ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+ free(my_data);
+ my_data = NULL;
+ return NULL;
+ }
- while (!adev->mixer && retry_num < RETRY_NUMBER) {
+ snd_card_num = 0;
usleep(RETRY_US);
- adev->mixer = mixer_open(snd_card_num);
- retry_num++;
+ continue;
}
+ if (card_verifed[snd_card_num]) {
+ ++snd_card_num;
+ continue;
+ }
+
+ adev->mixer = mixer_open(snd_card_num);
+
if (!adev->mixer) {
ALOGE("%s: Unable to open the mixer card: %d", __func__,
- snd_card_num);
- retry_num = 0;
- snd_card_num++;
+ snd_card_num);
+ ++snd_card_num;
continue;
}
+ card_verifed[snd_card_num] = true;
+
snd_card_name = mixer_get_name(adev->mixer);
ALOGV("%s: snd_card_name: %s", __func__, snd_card_name);
@@ -864,21 +891,21 @@
if (!adev->audio_route) {
ALOGE("%s: Failed to init audio route controls, aborting.",
__func__);
+ hw_info_deinit(my_data->hw_info);
+ my_data->hw_info = NULL;
free(my_data);
+ my_data = NULL;
+ mixer_close(adev->mixer);
+ adev->mixer = NULL;
return NULL;
}
adev->snd_card = snd_card_num;
ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
break;
}
- retry_num = 0;
- snd_card_num++;
- }
-
- if (snd_card_num >= MAX_SND_CARD) {
- ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
- free(my_data);
- return NULL;
+ ++snd_card_num;
+ mixer_close(adev->mixer);
+ adev->mixer = NULL;
}
//set max volume step for voice call
@@ -942,6 +969,12 @@
ALOGE("%s: Could not find the symbol acdb_loader_deallocate_ACDB from %s",
__func__, LIB_ACDB_LOADER);
+ my_data->acdb_send_audio_cal_v3 = (acdb_send_audio_cal_t)dlsym(my_data->acdb_handle,
+ "acdb_loader_send_audio_cal_v3");
+ if (!my_data->acdb_send_audio_cal_v3)
+ ALOGE("%s: Could not find the symbol acdb_send_audio_cal from %s",
+ __func__, LIB_ACDB_LOADER);
+
my_data->acdb_send_audio_cal = (acdb_send_audio_cal_t)dlsym(my_data->acdb_handle,
"acdb_loader_send_audio_cal");
if (!my_data->acdb_send_audio_cal)
@@ -1226,6 +1259,20 @@
return ret;
}
+int platform_get_default_app_type_v2(void *platform, usecase_type_t type, int *app_type)
+{
+ ALOGV("%s: platform: %p, type: %d", __func__, platform, type);
+ int rc = 0;
+ if (type == PCM_CAPTURE) {
+ *app_type = DEFAULT_APP_TYPE_TX_PATH;
+ } else if (type == PCM_PLAYBACK) {
+ *app_type = DEFAULT_APP_TYPE_RX_PATH;
+ } else {
+ rc = -EINVAL;
+ }
+ return rc;
+}
+
int platform_get_snd_device_acdb_id(snd_device_t snd_device)
{
if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
@@ -1243,6 +1290,7 @@
{
struct platform_data *my_data = (struct platform_data *)platform;
int acdb_dev_id, acdb_dev_type;
+ int sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
acdb_dev_id = acdb_device_table[audio_extn_get_spkr_prot_snd_device(snd_device)];
if (acdb_dev_id < 0) {
@@ -1250,14 +1298,32 @@
__func__, snd_device);
return -EINVAL;
}
- if (my_data->acdb_send_audio_cal) {
ALOGV("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
__func__, snd_device, acdb_dev_id);
- if (snd_device >= SND_DEVICE_OUT_BEGIN &&
- snd_device < SND_DEVICE_OUT_END)
- acdb_dev_type = ACDB_DEV_TYPE_OUT;
- else
- acdb_dev_type = ACDB_DEV_TYPE_IN;
+ if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END)
+ acdb_dev_type = ACDB_DEV_TYPE_OUT;
+ else
+ acdb_dev_type = ACDB_DEV_TYPE_IN;
+
+ if ((my_data->acdb_send_audio_cal_v3) &&
+ (snd_device == SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP) &&
+ !audio_extn_tfa_98xx_is_supported() ) {
+ /* TX path calibration */
+ my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_IN,
+ DEFAULT_APP_TYPE_TX_PATH, sample_rate, BUFF_IDX_0);
+ my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_OUT,
+ DEFAULT_APP_TYPE_RX_PATH, sample_rate, BUFF_IDX_0);
+ } else if ((my_data->acdb_send_audio_cal_v3) &&
+ (snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_HFP) &&
+ !audio_extn_tfa_98xx_is_supported()) {
+ /* RX path calibration */
+ ALOGV("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
+ __func__, snd_device, acdb_dev_id);
+ my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_IN,
+ DEFAULT_APP_TYPE_TX_PATH, sample_rate, BUFF_IDX_1);
+ my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_OUT,
+ DEFAULT_APP_TYPE_RX_PATH, sample_rate, BUFF_IDX_1);
+ } else if (my_data->acdb_send_audio_cal) {
my_data->acdb_send_audio_cal(acdb_dev_id, acdb_dev_type);
}
return 0;
@@ -1372,7 +1438,7 @@
DEFAULT_MUTE_RAMP_DURATION_MS};
if (audio_extn_hfp_is_active(adev))
- mixer_ctl_name = "HFP Tx Mute";
+ mixer_ctl_name = "HFP TX Mute";
set_values[0] = state;
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
@@ -1382,7 +1448,11 @@
return -EINVAL;
}
ALOGV("Setting voice mute state: %d", state);
- ret = mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+ // "HFP TX mute" mixer control has xcount of 1.
+ if (audio_extn_hfp_is_active(adev))
+ ret = mixer_ctl_set_array(ctl, set_values, 1 /*count*/);
+ else
+ ret = mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
return ret;
}
@@ -1503,7 +1573,7 @@
goto exit;
}
- if (mode == AUDIO_MODE_IN_CALL) {
+ if (mode == AUDIO_MODE_IN_CALL || audio_extn_hfp_is_active(adev)) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
devices & AUDIO_DEVICE_OUT_LINE) {
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 786e10f..fbc7e5f 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -421,6 +421,13 @@
return -ENODEV;
}
+int platform_get_default_app_type_v2(void *platform, usecase_type_t type __unused,
+ int *app_type __unused)
+{
+ ALOGE("%s: Not implemented", __func__);
+ return -ENOSYS;
+}
+
int platform_get_snd_device_acdb_id(snd_device_t snd_device __unused)
{
ALOGE("%s: Not implemented", __func__);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a5cfc18..9d4abec 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -510,6 +510,8 @@
{TO_NAME_INDEX(USECASE_INCALL_REC_DOWNLINK)},
{TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
{TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
+ {TO_NAME_INDEX(USECASE_VOICEMMODE1_CALL)},
+ {TO_NAME_INDEX(USECASE_VOICEMMODE2_CALL)},
};
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
@@ -562,7 +564,7 @@
char mccmnc[PROPERTY_VALUE_MAX];
char *ret = NULL;
- property_get("gsm.sim.operator.numeric",mccmnc,"0");
+ property_get("gsm.sim.operator.numeric",mccmnc,"00000");
list_for_each(node, &operator_info_list) {
info_item = node_to_item(node, struct operator_info, list);
@@ -1525,6 +1527,13 @@
return ret;
}
+int platform_get_default_app_type_v2(void *platform, usecase_type_t type __unused,
+ int *app_type)
+{
+ ALOGE("%s: Not implemented", __func__);
+ return -ENOSYS;
+}
+
int platform_get_snd_device_acdb_id(snd_device_t snd_device)
{
if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
@@ -1953,7 +1962,7 @@
goto exit;
}
- if (voice_is_in_call(adev) || adev->enable_voicerx) {
+ if (voice_is_in_call(adev) || adev->enable_voicerx || audio_extn_hfp_is_active(adev)) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
devices & AUDIO_DEVICE_OUT_LINE) {
diff --git a/hal/platform_api.h b/hal/platform_api.h
index fbc84c1..ebbe9d2 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -19,6 +19,7 @@
#include "voice.h"
#define MAX_VOLUME_CAL_STEPS 15
+#define CODEC_BACKEND_DEFAULT_SAMPLE_RATE 48000
struct amp_db_and_gain_table {
float amp;
@@ -27,6 +28,7 @@
};
enum card_status_t;
+enum usecase_type_t;
void *platform_init(struct audio_device *adev);
void platform_deinit(void *platform);
@@ -41,6 +43,7 @@
int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id);
int platform_get_snd_device_acdb_id(snd_device_t snd_device);
int platform_send_audio_calibration(void *platform, snd_device_t snd_device);
+int platform_get_default_app_type_v2(void *platform, enum usecase_type_t type, int *app_type);
int platform_switch_voice_call_device_pre(void *platform);
int platform_switch_voice_call_enable_device_config(void *platform,
snd_device_t out_snd_device,
diff --git a/hal/voice.c b/hal/voice.c
index d2215b6..611d875 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -29,6 +29,7 @@
#include "voice_extn/voice_extn.h"
#include "platform.h"
#include "platform_api.h"
+#include "audio_extn/tfa_98xx.h"
struct pcm_config pcm_config_voice_call = {
.channels = 1,
@@ -146,6 +147,11 @@
disable_snd_device(adev, uc_info->out_snd_device);
disable_snd_device(adev, uc_info->in_snd_device);
+ if (audio_extn_tfa_98xx_is_supported() && voice_get_mic_mute(adev)) {
+ voice_set_mic_mute(adev, false);
+ ALOGD("%s: unMute voice Tx", __func__);
+ }
+
list_remove(&uc_info->list);
free(uc_info);
@@ -211,6 +217,8 @@
pcm_start(session->pcm_tx);
pcm_start(session->pcm_rx);
+ audio_extn_tfa_98xx_enable_speaker();
+
/* Enable sidetone only when no calls are already active */
if (!voice_is_call_state_active(adev))
voice_set_sidetone(adev, uc_info->out_snd_device, true);
@@ -531,6 +539,7 @@
use_case_table[usecase->id]);
usecase->stream.out = adev->current_call_output;
select_devices(adev, usecase->id);
+ audio_extn_tfa_98xx_update();
}
}
}
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 59ee105..83f6a49 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msm8909,$(TARGET_BOARD_PLATFORM)),)
LOCAL_PATH:= $(call my-dir)
@@ -33,7 +33,7 @@
################################################################################
-ifneq ($(filter msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8909,$(TARGET_BOARD_PLATFORM)),)
include $(CLEAR_VARS)
diff --git a/post_proc/volume_listener.c b/post_proc/volume_listener.c
index 4a48a25..1402ae6 100644
--- a/post_proc/volume_listener.c
+++ b/post_proc/volume_listener.c
@@ -18,6 +18,7 @@
//#define LOG_NDEBUG 0
#include <stdlib.h>
#include <dlfcn.h>
+#include <math.h>
#include <cutils/list.h>
#include <cutils/log.h>
@@ -229,7 +230,7 @@
{
// iterate through list and make decision to set new gain dep cal level for speaker device
// 1. find all usecase active on speaker
- // 2. find average of left and right for each usecase
+ // 2. find energy sum for each usecase
// 3. find the highest of all the active usecase
// 4. if new value is different than the current value then load new calibration
@@ -243,15 +244,22 @@
ALOGV("%s ==> Start ...", __func__);
- // select the highest volume on speaker device
+ float sum_energy = 0.0;
+ bool sum_energy_used = false;
+ float temp_vol = 0;
+ // compute energy sum for the active speaker device (pick loudest of both channels)
list_for_each(node, &vol_effect_list) {
context = node_to_item(node, struct vol_listener_context_s, effect_list_node);
if ((context->state == VOL_LISTENER_STATE_ACTIVE) &&
- (context->dev_id & AUDIO_DEVICE_OUT_SPEAKER) &&
- (new_vol < (context->left_vol + context->right_vol) / 2)) {
- new_vol = (context->left_vol + context->right_vol) / 2;
+ (context->dev_id & AUDIO_DEVICE_OUT_SPEAKER)) {
+ sum_energy_used = true;
+ temp_vol = fmax(context->left_vol, context->right_vol);
+ sum_energy += temp_vol * temp_vol;
}
}
+ if (sum_energy_used) {
+ new_vol = fmin(sqrt(sum_energy), 1.0);
+ }
if (new_vol != current_vol) {
ALOGV("%s:: Change in decision :: current volume is %f new volume is %f",