Merge "hal: Add 8916 qrd skui sound card detection"
diff --git a/hal/Android.mk b/hal/Android.mk
index d3d909d..87ebeea 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -41,6 +41,10 @@
LOCAL_CFLAGS += -DANC_HEADSET_ENABLED
endif
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_FLUENCE)),true)
+ LOCAL_CFLAGS += -DFLUENCE_ENABLED
+endif
+
ifneq ($(strip $(AUDIO_FEATURE_DISABLED_PROXY_DEVICE)),true)
LOCAL_CFLAGS += -DAFE_PROXY_ENABLED
endif
@@ -86,7 +90,7 @@
endif
ifneq ($(strip, $(AUDIO_FEATURE_DISABLED_SPKR_PROTECTION)),true)
-ifneq ($(filter msm8974,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 apq8084,$(TARGET_BOARD_PLATFORM)),)
LOCAL_CFLAGS += -DSPKR_PROT_ENABLED
LOCAL_SRC_FILES += audio_extn/spkr_protection.c
LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 31a6353..d0caccb 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -28,6 +28,8 @@
#include "audio_hw.h"
#include "audio_extn.h"
+#include "platform.h"
+#include "platform_api.h"
#define MAX_SLEEP_RETRY 100
#define WIFI_INIT_WAIT_SLEEP 50
@@ -173,6 +175,60 @@
}
#endif /* ANC_HEADSET_ENABLED */
+#ifndef FLUENCE_ENABLED
+#define audio_extn_set_fluence_parameters(adev, parms) (0)
+#define audio_extn_get_fluence_parameters(adev, query, reply) (0)
+#else
+void audio_extn_set_fluence_parameters(struct audio_device *adev,
+ struct str_parms *parms)
+{
+ int ret = 0, err;
+ char value[32];
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE,
+ value, sizeof(value));
+ ALOGV_IF(err >= 0, "%s: Set Fluence Type to %s", __func__, value);
+ if (err >= 0) {
+ ret = platform_set_fluence_type(adev->platform, value);
+ if (ret != 0) {
+ ALOGE("platform_set_fluence_type returned error: %d", ret);
+ } else {
+ /*
+ *If the fluence is manually set/reset, devices
+ *need to get updated for all the usecases
+ *i.e. audio and voice.
+ */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ select_devices(adev, usecase->id);
+ }
+ }
+ }
+}
+
+int audio_extn_get_fluence_parameters(struct audio_device *adev,
+ struct str_parms *query, struct str_parms *reply)
+{
+ int ret = 0, err;
+ char value[256] = {0};
+
+ err = str_parms_get_str(query, AUDIO_PARAMETER_KEY_FLUENCE, value,
+ sizeof(value));
+ if (err >= 0) {
+ ret = platform_get_fluence_type(adev->platform, value, sizeof(value));
+ if (ret >= 0) {
+ ALOGV("%s: Fluence Type is %s", __func__, value);
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_FLUENCE, value);
+ } else
+ goto done;
+ }
+done:
+ return ret;
+}
+#endif /* FLUENCE_ENABLED */
+
#ifndef AFE_PROXY_ENABLED
#define audio_extn_set_afe_proxy_parameters(adev, parms) (0)
#define audio_extn_get_afe_proxy_parameters(query, reply) (0)
@@ -372,6 +428,7 @@
struct str_parms *parms)
{
audio_extn_set_anc_parameters(adev, parms);
+ audio_extn_set_fluence_parameters(adev, parms);
audio_extn_set_afe_proxy_parameters(adev, parms);
audio_extn_fm_set_parameters(adev, parms);
audio_extn_listen_set_parameters(adev, parms);
@@ -386,6 +443,7 @@
{
char *kv_pairs = NULL;
audio_extn_get_afe_proxy_parameters(query, reply);
+ audio_extn_get_fluence_parameters(adev, 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/audio_extn.h b/hal/audio_extn/audio_extn.h
index f631653..7feeb42 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -39,6 +39,16 @@
bool audio_extn_should_use_handset_anc(int in_channels);
#endif
+#ifndef FLUENCE_ENABLED
+#define audio_extn_set_fluence_parameters(adev, parms) (0)
+#define audio_extn_get_fluence_parameters(adev, query, reply) (0)
+#else
+void audio_extn_set_fluence_parameters(struct audio_device *adev,
+ struct str_parms *parms);
+int audio_extn_get_fluence_parameters(struct audio_device *adev,
+ struct str_parms *query, struct str_parms *reply);
+#endif
+
#ifndef AFE_PROXY_ENABLED
#define audio_extn_set_afe_proxy_channel_mixer(adev,channel_count) (0)
#define audio_extn_read_afe_proxy_channel_masks(out) (0)
@@ -61,6 +71,7 @@
#define audio_extn_usb_set_proxy_sound_card(sndcard_idx) (0)
#define audio_extn_usb_is_proxy_inuse() (0)
#else
+void initPlaybackVolume();
void audio_extn_usb_init(void *adev);
void audio_extn_usb_deinit();
void audio_extn_usb_start_playback(void *adev);
diff --git a/hal/audio_extn/fm.c b/hal/audio_extn/fm.c
index a4157f8..35b20b8 100644
--- a/hal/audio_extn/fm.c
+++ b/hal/audio_extn/fm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -64,7 +64,7 @@
{
int32_t vol, ret = 0;
struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "Internal FM RX Volume";
+ const char *mixer_ctl_name = FM_RX_VOLUME;
ALOGV("%s: entry", __func__);
ALOGD("%s: (%f)\n", __func__, value);
@@ -92,7 +92,6 @@
return -EINVAL;
}
mixer_ctl_set_value(ctl, 0, vol);
-
ALOGV("%s: exit", __func__);
return ret;
}
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index c480490..ed253c2 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -43,6 +43,7 @@
#ifdef HFP_ENABLED
#define AUDIO_PARAMETER_HFP_ENABLE "hfp_enable"
#define AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
+#define AUDIO_PARAMETER_KEY_HFP_VOLUME "hfp_volume"
static int32_t start_hfp(struct audio_device *adev,
struct str_parms *parms);
@@ -55,7 +56,7 @@
struct pcm *hfp_pcm_rx;
struct pcm *hfp_pcm_tx;
bool is_hfp_running;
- int hfp_volume;
+ float hfp_volume;
audio_usecase_t ucid;
};
@@ -79,6 +80,46 @@
.avail_min = 0,
};
+static int32_t hfp_set_volume(struct audio_device *adev, float value)
+{
+ int32_t vol, ret = 0;
+ struct mixer_ctl *ctl;
+ const char *mixer_ctl_name = "Internal HFP RX Volume";
+
+ ALOGV("%s: entry", __func__);
+ ALOGD("%s: (%f)\n", __func__, value);
+
+ if (value < 0.0) {
+ ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
+ value = 0.0;
+ } else {
+ value = ((value > 15.000000) ? 1.0 : (value / 15));
+ 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__);
+ return -EIO;
+ }
+
+ ALOGD("%s: Setting HFP volume to %d \n", __func__, vol);
+ 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;
+ }
+ if(mixer_ctl_set_value(ctl, 0, vol) < 0) {
+ ALOGE("%s: Couldn't set HFP Volume: [%d]", __func__, vol);
+ return -EINVAL;
+ }
+
+ ALOGV("%s: exit", __func__);
+ return ret;
+}
+
static int32_t start_hfp(struct audio_device *adev,
struct str_parms *parms)
{
@@ -158,8 +199,8 @@
pcm_start(hfpmod.hfp_pcm_rx);
pcm_start(hfpmod.hfp_pcm_tx);
-
hfpmod.is_hfp_running = true;
+ hfp_set_volume(adev, hfpmod.hfp_volume);
ALOGD("%s: exit: status(%d)", __func__, ret);
return 0;
@@ -238,6 +279,7 @@
int ret;
int rate;
int val;
+ float vol;
char value[32]={0};
ret = str_parms_get_str(parms, AUDIO_PARAMETER_HFP_ENABLE, value,
@@ -275,5 +317,20 @@
select_devices(adev, hfpmod.ucid);
}
}
+
+ memset(value, 0, sizeof(value));
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_HFP_VOLUME,
+ value, sizeof(value));
+ if (ret >= 0) {
+ if (sscanf(value, "%f", &vol) != 1){
+ ALOGE("%s: error in retrieving hfp volume", __func__);
+ ret = -EIO;
+ goto exit;
+ }
+ ALOGD("%s: set_hfp_volume usecase, Vol: [%f]", __func__, vol);
+ hfp_set_volume(adev, vol);
+ }
+exit:
+ ALOGV("%s Exit",__func__);
}
#endif /*HFP_ENABLED*/
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 699c3b7..281b445 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -96,6 +96,42 @@
usbmod = calloc(1, sizeof(struct usb_module));
}
+// Some USB audio accessories have a really low default volume set. Look for a suitable
+// volume control and set the volume to default volume level.
+static void initPlaybackVolume() {
+ ALOGD("initPlaybackVolume");
+ struct mixer *usbMixer = mixer_open(1);
+
+ if (usbMixer) {
+ struct mixer_ctl *ctl = NULL;
+ unsigned int usbPlaybackVolume;
+ unsigned int i;
+ unsigned int num_ctls = mixer_get_num_ctls(usbMixer);
+
+ // Look for the first control named ".*Playback Volume" that isn't for a microphone
+ for (i = 0; i < num_ctls; i++) {
+ ctl = mixer_get_ctl(usbMixer, i);
+ if (strstr((const char *)mixer_ctl_get_name(ctl), "Playback Volume") &&
+ !strstr((const char *)mixer_ctl_get_name(ctl), "Mic")) {
+ break;
+ }
+ }
+ if (ctl != NULL) {
+ ALOGD("Found a volume control for USB: %s", mixer_ctl_get_name(ctl) );
+ usbPlaybackVolume = mixer_ctl_get_value(ctl, 0);
+ ALOGD("Value got from mixer_ctl_get is:%u", usbPlaybackVolume);
+ if (mixer_ctl_set_value(ctl,0,usbPlaybackVolume) < 0) {
+ ALOGE("Failed to set volume; default volume might be used");
+ }
+ } else {
+ ALOGE("No playback volume control found; default volume will be used");
+ }
+ mixer_close(usbMixer);
+ } else {
+ ALOGE("Failed to open mixer for card 1");
+ }
+}
+
static int usb_get_numof_rates(char *rates_str)
{
int i, size = 0;
@@ -340,7 +376,7 @@
pcm_config_usbmod.channels = usbmod->channels_playback;
pcm_config_usbmod.period_count = AFE_PROXY_PERIOD_COUNT;
usbmod->proxy_device_id = AFE_PROXY_PLAYBACK_DEVICE;
- ALOGV("%s: proxy device %u:period %u:channels %u:sample", __func__,
+ ALOGD("%s: proxy device %u:period %u:channels %u:sample", __func__,
pcm_config_usbmod.period_size, pcm_config_usbmod.channels,
pcm_config_usbmod.rate);
@@ -375,6 +411,8 @@
ALOGD("%s: PROXY configured for playback", __func__);
pthread_mutex_unlock(&usbmod->usb_playback_lock);
+ ALOGD("Init USB volume");
+ initPlaybackVolume();
/* main loop to read from proxy and write to usb */
while (usbmod->is_playback_running) {
/* read data from proxy */
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 69e9561..2f67784 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -296,6 +296,20 @@
return 0;
}
+ /* Set BT sample rate before enabling the devices. Adding sample rate mixer
+ * control in use-case does not work because rate update takes place after
+ * AFE port open due to the limitation of mixer control order execution.
+ */
+ if ((snd_device == SND_DEVICE_OUT_BT_SCO) ||
+ (snd_device == SND_DEVICE_IN_BT_SCO_MIC)) {
+ audio_route_apply_path(adev->audio_route, BT_SCO_SAMPLE_RATE);
+ audio_route_update_mixer(adev->audio_route);
+ } else if ((snd_device == SND_DEVICE_OUT_BT_SCO_WB) ||
+ (snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB)) {
+ audio_route_apply_path(adev->audio_route, BT_SCO_WB_SAMPLE_RATE);
+ audio_route_update_mixer(adev->audio_route);
+ }
+
/* start usb playback thread */
if(SND_DEVICE_OUT_USB_HEADSET == snd_device ||
SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET == snd_device)
@@ -305,13 +319,14 @@
if(SND_DEVICE_IN_USB_HEADSET_MIC == snd_device)
audio_extn_usb_start_capture(adev);
- if (snd_device == SND_DEVICE_OUT_SPEAKER &&
+ if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
audio_extn_spkr_prot_is_enabled()) {
- if (audio_extn_spkr_prot_start_processing(snd_device)) {
- ALOGE("%s: spkr_start_processing failed", __func__);
- return -EINVAL;
- }
- } else {
+ if (audio_extn_spkr_prot_start_processing(snd_device)) {
+ ALOGE("%s: spkr_start_processing failed", __func__);
+ return -EINVAL;
+ }
+ } else {
ALOGV("%s: snd_device(%d: %s)", __func__,
snd_device, device_name);
if (platform_send_audio_calibration(adev->platform, snd_device) < 0) {
@@ -364,7 +379,8 @@
if(SND_DEVICE_IN_USB_HEADSET_MIC == snd_device)
audio_extn_usb_stop_capture(adev);
- if (snd_device == SND_DEVICE_OUT_SPEAKER &&
+ if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
audio_extn_spkr_prot_is_enabled()) {
audio_extn_spkr_prot_stop_processing();
} else
@@ -423,23 +439,22 @@
/* Make sure all the streams are de-routed before disabling the device */
audio_route_update_mixer(adev->audio_route);
+ /* Make sure the previous devices to be disabled first and then enable the
+ selected devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- disable_snd_device(adev, usecase->out_snd_device, false);
+ disable_snd_device(adev, usecase->out_snd_device, true);
}
}
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- enable_snd_device(adev, snd_device, false);
+ enable_snd_device(adev, snd_device, true);
}
}
- /* Make sure new snd device is enabled before re-routing the streams */
- audio_route_update_mixer(adev->audio_route);
-
/* Re-route all the usecases on the shared backend other than the
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
@@ -493,23 +508,22 @@
/* Make sure all the streams are de-routed before disabling the device */
audio_route_update_mixer(adev->audio_route);
+ /* Make sure the previous devices to be disabled first and then enable the
+ selected devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- disable_snd_device(adev, usecase->in_snd_device, false);
+ disable_snd_device(adev, usecase->in_snd_device, true);
}
}
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- enable_snd_device(adev, snd_device, false);
+ enable_snd_device(adev, snd_device, true);
}
}
- /* Make sure new snd device is enabled before re-routing the streams */
- audio_route_update_mixer(adev->audio_route);
-
/* Re-route all the usecases on the shared backend other than the
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
@@ -551,21 +565,6 @@
return ret;
}
-static void update_devices_for_all_voice_usecases(struct audio_device *adev)
-{
- struct listnode *node;
- struct audio_usecase *usecase;
-
- list_for_each(node, &adev->usecase_list) {
- usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == VOICE_CALL) {
- ALOGV("%s: updating device for usecase:%s", __func__,
- use_case_table[usecase->id]);
- select_devices(adev, usecase->id);
- }
- }
-}
-
static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
{
struct audio_usecase *usecase;
@@ -638,7 +637,8 @@
}
} else if (voice_extn_compress_voip_is_active(adev)) {
voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
- if (voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
+ if ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) {
in_snd_device = voip_usecase->in_snd_device;
out_snd_device = voip_usecase->out_snd_device;
}
@@ -699,12 +699,12 @@
/* Disable current sound devices */
if (usecase->out_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase, true);
- disable_snd_device(adev, usecase->out_snd_device, false);
+ disable_snd_device(adev, usecase->out_snd_device, true);
}
if (usecase->in_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase, true);
- disable_snd_device(adev, usecase->in_snd_device, false);
+ disable_snd_device(adev, usecase->in_snd_device, true);
}
/* Applicable only on the targets that has external modem.
@@ -1425,13 +1425,14 @@
pthread_mutex_lock(&adev->lock);
/*
- * When HDMI cable is unplugged the music playback is paused and
- * the policy manager sends routing=0. But the audioflinger
- * continues to write data until standby time (3sec).
- * As the HDMI core is turned off, the write gets blocked.
+ * When HDMI cable is unplugged/usb hs is disconnected the
+ * music playback is paused and the policy manager sends routing=0
+ * But the audioflingercontinues to write data until standby time
+ * (3sec). As the HDMI core is turned off, the write gets blocked.
* Avoid this by routing audio to speaker until standby.
*/
- if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL &&
+ if ((out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+ out->devices == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET) &&
val == AUDIO_DEVICE_NONE) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
@@ -1467,7 +1468,7 @@
} else if ((adev->mode == AUDIO_MODE_IN_CALL) &&
voice_is_in_call(adev) &&
(out == adev->primary_output)) {
- update_devices_for_all_voice_usecases(adev);
+ voice_update_devices_for_all_voice_usecases(adev);
}
}
@@ -2256,7 +2257,8 @@
/* Check if this usecase is already existing */
pthread_mutex_lock(&adev->lock);
- if (get_usecase_from_list(adev, out->usecase) != NULL) {
+ if ((get_usecase_from_list(adev, out->usecase) != NULL) &&
+ (out->usecase != USECASE_COMPRESS_VOIP_CALL)) {
ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
pthread_mutex_unlock(&adev->lock);
ret = -EEXIST;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 2108a00..4a32602 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -31,14 +31,20 @@
#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"
+#define BT_SCO_SAMPLE_RATE "bt-sco-samplerate"
+#define BT_SCO_WB_SAMPLE_RATE "bt-sco-wb-samplerate"
+
/* Flags used to initialize acdb_settings variable that goes to ACDB library */
-#define DMIC_FLAG 0x00000002
-#define QMIC_FLAG 0x00000004
-#define TTY_MODE_OFF 0x00000010
-#define TTY_MODE_FULL 0x00000020
-#define TTY_MODE_VCO 0x00000040
-#define TTY_MODE_HCO 0x00000080
-#define TTY_MODE_CLEAR 0xFFFFFF0F
+#define NONE_FLAG 0x00000000
+#define ANC_FLAG 0x00000001
+#define DMIC_FLAG 0x00000002
+#define QMIC_FLAG 0x00000004
+#define TTY_MODE_OFF 0x00000010
+#define TTY_MODE_FULL 0x00000020
+#define TTY_MODE_VCO 0x00000040
+#define TTY_MODE_HCO 0x00000080
+#define TTY_MODE_CLEAR 0xFFFFFF0F
+#define FLUENCE_MODE_CLEAR 0xFFFFFFF0
#define ACDB_DEV_TYPE_OUT 1
#define ACDB_DEV_TYPE_IN 2
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 70e8c55..c6ba144 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -105,6 +105,7 @@
bool fluence_in_voice_rec;
bool fluence_in_audio_rec;
int fluence_type;
+ char fluence_cap[PROPERTY_VALUE_MAX];
int btsco_sample_rate;
bool slowtalk;
/* Audio calibration related functions */
@@ -142,6 +143,7 @@
[USECASE_VOICE2_CALL] = {VOICE2_CALL_PCM_DEVICE, VOICE2_CALL_PCM_DEVICE},
[USECASE_VOLTE_CALL] = {VOLTE_CALL_PCM_DEVICE, VOLTE_CALL_PCM_DEVICE},
[USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE, QCHAT_CALL_PCM_DEVICE},
+ [USECASE_VOWLAN_CALL] = {VOWLAN_CALL_PCM_DEVICE, VOWLAN_CALL_PCM_DEVICE},
[USECASE_COMPRESS_VOIP_CALL] = {COMPRESS_VOIP_CALL_PCM_DEVICE, COMPRESS_VOIP_CALL_PCM_DEVICE},
[USECASE_INCALL_REC_UPLINK] = {AUDIO_RECORD_PCM_DEVICE,
AUDIO_RECORD_PCM_DEVICE},
@@ -605,10 +607,10 @@
my_data->fluence_in_audio_rec = false;
my_data->fluence_type = FLUENCE_NONE;
- property_get("ro.qc.sdk.audio.fluencetype", value, "");
- if (!strncmp("fluencepro", value, sizeof("fluencepro"))) {
+ property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
+ if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
my_data->fluence_type = FLUENCE_QUAD_MIC | FLUENCE_DUAL_MIC;
- } else if (!strncmp("fluence", value, sizeof("fluence"))) {
+ } else if (!strncmp("fluence", my_data->fluence_cap, sizeof("fluence"))) {
my_data->fluence_type = FLUENCE_DUAL_MIC;
} else {
my_data->fluence_type = FLUENCE_NONE;
@@ -787,6 +789,63 @@
return ret;
}
+int platform_set_fluence_type(void *platform, char *value)
+{
+ int ret = 0;
+ int fluence_type = FLUENCE_NONE;
+ int fluence_flag = NONE_FLAG;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct audio_device *adev = my_data->adev;
+
+ ALOGV("%s: fluence type:%d", __func__, my_data->fluence_type);
+
+ /* only dual mic turn on and off is supported as of now through setparameters */
+ if (!strncmp(AUDIO_PARAMETER_VALUE_DUALMIC,value, sizeof(AUDIO_PARAMETER_VALUE_DUALMIC))) {
+ if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro")) ||
+ !strncmp("fluence", my_data->fluence_cap, sizeof("fluence"))) {
+ ALOGV("fluence dualmic feature enabled \n");
+ fluence_type = FLUENCE_DUAL_MIC;
+ fluence_flag = DMIC_FLAG;
+ } else {
+ ALOGE("%s: Failed to set DUALMIC", __func__);
+ ret = -1;
+ goto done;
+ }
+ } else if (!strncmp(AUDIO_PARAMETER_KEY_NO_FLUENCE, value, sizeof(AUDIO_PARAMETER_KEY_NO_FLUENCE))) {
+ ALOGV("fluence disabled");
+ fluence_type = FLUENCE_NONE;
+ } else {
+ ALOGE("Invalid fluence value : %s",value);
+ ret = -1;
+ goto done;
+ }
+
+ if (fluence_type != my_data->fluence_type) {
+ ALOGV("%s: Updating fluence_type to :%d", __func__, fluence_type);
+ my_data->fluence_type = fluence_type;
+ adev->acdb_settings = (adev->acdb_settings & FLUENCE_MODE_CLEAR) | fluence_flag;
+ }
+done:
+ return ret;
+}
+
+int platform_get_fluence_type(void *platform, char *value, uint32_t len)
+{
+ int ret = 0;
+ struct platform_data *my_data = (struct platform_data *)platform;
+
+ if (my_data->fluence_type == FLUENCE_QUAD_MIC) {
+ strlcpy(value, "quadmic", len);
+ } else if (my_data->fluence_type == FLUENCE_DUAL_MIC) {
+ strlcpy(value, "dualmic", len);
+ } else if (my_data->fluence_type == FLUENCE_NONE) {
+ strlcpy(value, "none", len);
+ } else
+ ret = -1;
+
+ return ret;
+}
+
int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
{
int ret = 0;
@@ -1255,6 +1314,7 @@
}
} else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
+ set_echo_reference(adev->mixer, EC_REF_RX);
} else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
@@ -1273,6 +1333,7 @@
}
} else {
snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
+ set_echo_reference(adev->mixer, EC_REF_RX);
}
}
} else if (source == AUDIO_SOURCE_CAMCORDER) {
@@ -1708,23 +1769,7 @@
char *str = NULL;
char value[256] = {0};
int ret;
- int fluence_type;
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_FLUENCE_TYPE,
- value, sizeof(value));
- if (ret >= 0) {
- if (my_data->fluence_type & FLUENCE_QUAD_MIC) {
- strlcpy(value, "fluencepro", sizeof(value));
- } else if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
- strlcpy(value, "fluence", sizeof(value));
- } else {
- strlcpy(value, "none", sizeof(value));
- }
-
- str_parms_add_str(reply, AUDIO_PARAMETER_KEY_FLUENCE_TYPE, value);
- }
-
- memset(value, 0, sizeof(value));
ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_SLOWTALK,
value, sizeof(value));
if (ret >= 0) {
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index cad5198..f12697c 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -168,8 +168,8 @@
#define FM_PLAYBACK_PCM_DEVICE 5
#define FM_CAPTURE_PCM_DEVICE 6
#define HFP_PCM_RX 5
-#define HFP_SCO_RX 23
-#define HFP_ASM_RX_TX 24
+#define HFP_SCO_RX 17
+#define HFP_ASM_RX_TX 18
#define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
@@ -178,6 +178,8 @@
#define PLAYBACK_OFFLOAD_DEVICE 9
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
+/* Define macro for Internal FM volume mixer */
+#define FM_RX_VOLUME "Internal FM RX Volume"
#define LOWLATENCY_PCM_DEVICE 12
#define EC_REF_RX "I2S_RX"
@@ -187,6 +189,7 @@
#define VOICE2_CALL_PCM_DEVICE 13
#define VOLTE_CALL_PCM_DEVICE 15
#define QCHAT_CALL_PCM_DEVICE 14
+#define VOWLAN_CALL_PCM_DEVICE 16
#define LIB_CSD_CLIENT "libcsd-client.so"
/* CSD-CLIENT related functions */
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index e38d801..e326785 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -118,4 +118,7 @@
#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
#define AUDIO_CAPTURE_PERIOD_COUNT 2
+/* Define macro for Internal FM volume mixer */
+#define FM_RX_VOLUME "Internal FM RX Volume"
+
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 6920efe..f43a3b0 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -92,6 +92,11 @@
SND_DEVICE_IN_QUAD_MIC,
};
+static const snd_device_t taiko_apq8084_sbc_variant_devices[] = {
+ SND_DEVICE_IN_HANDSET_MIC,
+ SND_DEVICE_IN_SPEAKER_MIC,
+};
+
static const snd_device_t tapan_lite_variant_devices[] = {
SND_DEVICE_OUT_SPEAKER,
SND_DEVICE_OUT_HEADPHONES,
@@ -157,6 +162,12 @@
hw_info->snd_devices = (snd_device_t *)taiko_liquid_variant_devices;
hw_info->num_snd_devices = ARRAY_SIZE(taiko_liquid_variant_devices);
strlcpy(hw_info->dev_extn, "-liquid", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "apq8084-taiko-sbc-snd-card")) {
+ strlcpy(hw_info->type, " sbc", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
+ hw_info->snd_devices = (snd_device_t *)taiko_apq8084_sbc_variant_devices;
+ hw_info->num_snd_devices = ARRAY_SIZE(taiko_apq8084_sbc_variant_devices);
+ strlcpy(hw_info->dev_extn, "-sbc", sizeof(hw_info->dev_extn));
} else {
ALOGW("%s: Not an 8084 device", __func__);
}
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index eea048f..5315e78 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -113,6 +113,7 @@
bool fluence_in_audio_rec;
int fluence_type;
int fluence_mode;
+ char fluence_cap[PROPERTY_VALUE_MAX];
int btsco_sample_rate;
bool slowtalk;
bool is_i2s_ext_modem;
@@ -681,10 +682,10 @@
my_data->fluence_type = FLUENCE_NONE;
my_data->fluence_mode = FLUENCE_ENDFIRE;
- property_get("ro.qc.sdk.audio.fluencetype", value, "");
- if (!strncmp("fluencepro", value, sizeof("fluencepro"))) {
+ property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
+ if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
my_data->fluence_type = FLUENCE_QUAD_MIC | FLUENCE_DUAL_MIC;
- } else if (!strncmp("fluence", value, sizeof("fluence"))) {
+ } else if (!strncmp("fluence", my_data->fluence_cap, sizeof("fluence"))) {
my_data->fluence_type = FLUENCE_DUAL_MIC;
} else {
my_data->fluence_type = FLUENCE_NONE;
@@ -838,6 +839,13 @@
strlcat(mixer_path, " capture-fm", MIXER_PATH_MAX_LENGTH);
else if (snd_device == SND_DEVICE_OUT_TRANSMISSION_FM)
strlcat(mixer_path, " transmission-fm", MIXER_PATH_MAX_LENGTH);
+ else if (snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
+ audio_extn_spkr_prot_is_enabled() ) {
+ char platform[PROPERTY_VALUE_MAX];
+ property_get("ro.board.platform", platform, "");
+ if (!strncmp("apq8084", platform, sizeof("apq8084")))
+ strlcat(mixer_path, " speaker-protected", MIXER_PATH_MAX_LENGTH);
+ }
}
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type)
@@ -874,6 +882,63 @@
return ret;
}
+int platform_set_fluence_type(void *platform, char *value)
+{
+ int ret = 0;
+ int fluence_type = FLUENCE_NONE;
+ int fluence_flag = NONE_FLAG;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct audio_device *adev = my_data->adev;
+
+ ALOGV("%s: fluence type:%d", __func__, my_data->fluence_type);
+
+ /* only dual mic turn on and off is supported as of now through setparameters */
+ if (!strncmp(AUDIO_PARAMETER_VALUE_DUALMIC,value, sizeof(AUDIO_PARAMETER_VALUE_DUALMIC))) {
+ if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro")) ||
+ !strncmp("fluence", my_data->fluence_cap, sizeof("fluence"))) {
+ ALOGV("fluence dualmic feature enabled \n");
+ fluence_type = FLUENCE_DUAL_MIC;
+ fluence_flag = DMIC_FLAG;
+ } else {
+ ALOGE("%s: Failed to set DUALMIC", __func__);
+ ret = -1;
+ goto done;
+ }
+ } else if (!strncmp(AUDIO_PARAMETER_KEY_NO_FLUENCE, value, sizeof(AUDIO_PARAMETER_KEY_NO_FLUENCE))) {
+ ALOGV("fluence disabled");
+ fluence_type = FLUENCE_NONE;
+ } else {
+ ALOGE("Invalid fluence value : %s",value);
+ ret = -1;
+ goto done;
+ }
+
+ if (fluence_type != my_data->fluence_type) {
+ ALOGV("%s: Updating fluence_type to :%d", __func__, fluence_type);
+ my_data->fluence_type = fluence_type;
+ adev->acdb_settings = (adev->acdb_settings & FLUENCE_MODE_CLEAR) | fluence_flag;
+ }
+done:
+ return ret;
+}
+
+int platform_get_fluence_type(void *platform, char *value, uint32_t len)
+{
+ int ret = 0;
+ struct platform_data *my_data = (struct platform_data *)platform;
+
+ if (my_data->fluence_type == FLUENCE_QUAD_MIC) {
+ strlcpy(value, "quadmic", len);
+ } else if (my_data->fluence_type == FLUENCE_DUAL_MIC) {
+ strlcpy(value, "dualmic", len);
+ } else if (my_data->fluence_type == FLUENCE_NONE) {
+ strlcpy(value, "none", len);
+ } else
+ ret = -1;
+
+ return ret;
+}
+
int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
{
int ret = 0;
@@ -939,21 +1004,28 @@
int acdb_rx_id, acdb_tx_id;
int ret = 0;
- acdb_rx_id = acdb_device_table[out_snd_device];
+ if (my_data->csd == NULL)
+ return ret;
+
+ if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
+ audio_extn_spkr_prot_is_enabled())
+ acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
+ else
+ acdb_rx_id = acdb_device_table[out_snd_device];
+
acdb_tx_id = acdb_device_table[in_snd_device];
- if (my_data->csd != NULL) {
- if (acdb_rx_id > 0 && acdb_tx_id > 0) {
- ret = my_data->csd->enable_device_config(acdb_rx_id, acdb_tx_id);
- if (ret < 0) {
- ALOGE("%s: csd_enable_device_config, failed, error %d",
- __func__, ret);
- }
- } else {
- ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
- acdb_rx_id, acdb_tx_id);
+ if (acdb_rx_id > 0 && acdb_tx_id > 0) {
+ ret = my_data->csd->enable_device_config(acdb_rx_id, acdb_tx_id);
+ if (ret < 0) {
+ ALOGE("%s: csd_enable_device_config, failed, error %d",
+ __func__, ret);
}
+ } else {
+ ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
+ acdb_rx_id, acdb_tx_id);
}
+
return ret;
}
@@ -988,22 +1060,28 @@
int acdb_rx_id, acdb_tx_id;
int ret = 0;
- acdb_rx_id = acdb_device_table[out_snd_device];
+ if (my_data->csd == NULL)
+ return ret;
+
+ if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
+ audio_extn_spkr_prot_is_enabled())
+ acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED];
+ else
+ acdb_rx_id = acdb_device_table[out_snd_device];
+
acdb_tx_id = acdb_device_table[in_snd_device];
- if (my_data->csd != NULL) {
- if (acdb_rx_id > 0 && acdb_tx_id > 0) {
- ret = my_data->csd->enable_device(acdb_rx_id, acdb_tx_id,
- my_data->adev->acdb_settings);
- if (ret < 0) {
- ALOGE("%s: csd_enable_device, failed, error %d",
- __func__, ret);
- }
- } else {
- ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
- acdb_rx_id, acdb_tx_id);
+ if (acdb_rx_id > 0 && acdb_tx_id > 0) {
+ ret = my_data->csd->enable_device(acdb_rx_id, acdb_tx_id,
+ my_data->adev->acdb_settings);
+ if (ret < 0) {
+ ALOGE("%s: csd_enable_device, failed, error %d", __func__, ret);
}
+ } else {
+ ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
+ acdb_rx_id, acdb_tx_id);
}
+
return ret;
}
@@ -1076,7 +1154,8 @@
mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
if (my_data->csd != NULL) {
- ret = my_data->csd->volume(ALL_SESSION_VSID, volume);
+ ret = my_data->csd->volume(ALL_SESSION_VSID, volume,
+ DEFAULT_VOLUME_RAMP_DURATION_MS);
if (ret < 0) {
ALOGE("%s: csd_volume error %d", __func__, ret);
}
@@ -1093,7 +1172,7 @@
int ret = 0;
uint32_t set_values[ ] = {0,
ALL_SESSION_VSID,
- DEFAULT_VOLUME_RAMP_DURATION_MS};
+ DEFAULT_MUTE_RAMP_DURATION_MS};
set_values[0] = state;
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
@@ -1106,7 +1185,8 @@
mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
if (my_data->csd != NULL) {
- ret = my_data->csd->mic_mute(ALL_SESSION_VSID, state);
+ ret = my_data->csd->mic_mute(ALL_SESSION_VSID, state,
+ DEFAULT_MUTE_RAMP_DURATION_MS);
if (ret < 0) {
ALOGE("%s: csd_mic_mute error %d", __func__, ret);
}
@@ -1342,6 +1422,7 @@
if (out_device & AUDIO_DEVICE_OUT_EARPIECE &&
audio_extn_should_use_handset_anc(channel_count)) {
snd_device = SND_DEVICE_IN_AANC_HANDSET_MIC;
+ adev->acdb_settings |= ANC_FLAG;
} else if (my_data->fluence_type == FLUENCE_NONE ||
my_data->fluence_in_voice_call == false) {
snd_device = SND_DEVICE_IN_HANDSET_MIC;
@@ -1352,6 +1433,7 @@
}
} else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
+ set_echo_reference(adev->mixer, EC_REF_RX);
} else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
@@ -1373,6 +1455,7 @@
}
} else {
snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
+ set_echo_reference(adev->mixer, EC_REF_RX);
}
}
} else if (source == AUDIO_SOURCE_CAMCORDER) {
@@ -1687,11 +1770,6 @@
if (err >= 0) {
str_parms_del(parms, AUDIO_PARAMETER_KEY_BTSCO);
my_data->btsco_sample_rate = val;
- if (val == SAMPLE_RATE_16KHZ) {
- audio_route_apply_path(my_data->adev->audio_route,
- "bt-sco-wb-samplerate");
- audio_route_update_mixer(my_data->adev->audio_route);
- }
}
err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SLOWTALK, value, sizeof(value));
@@ -1824,24 +1902,8 @@
char *str = NULL;
char value[256] = {0};
int ret;
- int fluence_type;
char *kv_pairs = NULL;
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_FLUENCE_TYPE,
- value, sizeof(value));
- if (ret >= 0) {
- if (my_data->fluence_type & FLUENCE_QUAD_MIC) {
- strlcpy(value, "fluencepro", sizeof(value));
- } else if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
- strlcpy(value, "fluence", sizeof(value));
- } else {
- strlcpy(value, "none", sizeof(value));
- }
-
- str_parms_add_str(reply, AUDIO_PARAMETER_KEY_FLUENCE_TYPE, value);
- }
-
- memset(value, 0, sizeof(value));
ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_SLOWTALK,
value, sizeof(value));
if (ret >= 0) {
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index c22b2b8..209ee2b 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -139,7 +139,7 @@
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define ALL_SESSION_VSID 0xFFFFFFFF
-#define DEFAULT_MUTE_RAMP_DURATION 500
+#define DEFAULT_MUTE_RAMP_DURATION_MS 20
#define DEFAULT_VOLUME_RAMP_DURATION_MS 20
#define MIXER_PATH_MAX_LENGTH 100
@@ -180,9 +180,21 @@
#define HFP_PCM_RX 5
#define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
+
+#ifdef PLATFORM_MSM8610
+#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 14
+#elif PLATFORM_MSM8x26
#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
+#else
+#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 35
+#endif
+
#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
+#ifdef PLATFORM_APQ8084
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 33
+#else
#define SPKR_PROT_CALIB_TX_PCM_DEVICE 25
+#endif
#define PLAYBACK_OFFLOAD_DEVICE 9
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
@@ -207,9 +219,9 @@
#define VOWLAN_CALL_PCM_DEVICE 30
#elif PLATFORM_APQ8084
#define VOICE_CALL_PCM_DEVICE 20
-#define VOICE2_CALL_PCM_DEVICE 28
+#define VOICE2_CALL_PCM_DEVICE 25
#define VOLTE_CALL_PCM_DEVICE 21
-#define QCHAT_CALL_PCM_DEVICE 06
+#define QCHAT_CALL_PCM_DEVICE 33
#define VOWLAN_CALL_PCM_DEVICE -1
#elif PLATFORM_MSM8610
#define VOICE_CALL_PCM_DEVICE 2
@@ -233,6 +245,12 @@
#define HFP_ASM_RX_TX 24
#endif
+#ifdef PLATFORM_APQ8084
+#define FM_RX_VOLUME "Quat MI2S FM RX Volume"
+#else
+#define FM_RX_VOLUME "Internal FM RX Volume"
+#endif
+
#define LIB_CSD_CLIENT "libcsd-client.so"
/* CSD-CLIENT related functions */
typedef int (*init_t)(bool);
@@ -240,8 +258,8 @@
typedef int (*disable_device_t)();
typedef int (*enable_device_config_t)(int, int);
typedef int (*enable_device_t)(int, int, uint32_t);
-typedef int (*volume_t)(uint32_t, int);
-typedef int (*mic_mute_t)(uint32_t, int);
+typedef int (*volume_t)(uint32_t, int, uint16_t);
+typedef int (*mic_mute_t)(uint32_t, int, uint16_t);
typedef int (*slow_talk_t)(uint32_t, uint8_t);
typedef int (*start_voice_t)(uint32_t);
typedef int (*stop_voice_t)(uint32_t);
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 03890e9..bf6bdcb 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -28,6 +28,8 @@
void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
int platform_get_snd_device_index(char *snd_device_index_name);
+int platform_set_fluence_type(void *platform, char *value);
+int platform_get_fluence_type(void *platform, char *value, uint32_t len);
int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id);
int platform_send_audio_calibration(void *platform, snd_device_t snd_device);
int platform_switch_voice_call_device_pre(void *platform);
diff --git a/hal/voice.c b/hal/voice.c
index 1c3ab38..ac067a3 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -405,8 +405,7 @@
adev->voice.tty_mode = tty_mode;
adev->acdb_settings = (adev->acdb_settings & TTY_MODE_CLEAR) | tty_mode;
if (voice_is_in_call(adev))
- //todo: what about voice2, volte and qchat usecases?
- select_devices(adev, USECASE_VOICE_CALL);
+ voice_update_devices_for_all_voice_usecases(adev);
}
}
@@ -446,4 +445,19 @@
voice_extn_init(adev);
}
+void voice_update_devices_for_all_voice_usecases(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == VOICE_CALL) {
+ ALOGV("%s: updating device for usecase:%s", __func__,
+ use_case_table[usecase->id]);
+ select_devices(adev, usecase->id);
+ }
+ }
+}
+
diff --git a/hal/voice.h b/hal/voice.h
index 5e4ce7f..d160569 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -86,4 +86,5 @@
struct stream_out *out);
int voice_check_and_stop_incall_rec_usecase(struct audio_device *adev,
struct stream_in *in);
+void voice_update_devices_for_all_voice_usecases(struct audio_device *adev);
#endif //VOICE_H
diff --git a/mm-audio/aenc-aac/qdsp6/inc/aenc_svr.h b/mm-audio/aenc-aac/qdsp6/inc/aenc_svr.h
index 72bfebe..9bc7200 100644
--- a/mm-audio/aenc-aac/qdsp6/inc/aenc_svr.h
+++ b/mm-audio/aenc-aac/qdsp6/inc/aenc_svr.h
@@ -60,7 +60,7 @@
#endif
#define DEBUG_PRINT_ERROR LOGE
-#define DEBUG_PRINT LOGI
+#define DEBUG_PRINT LOGV
#define DEBUG_DETAIL LOGV
typedef void (*message_func)(void* client_data, unsigned char id);
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 8947456..a6f0b0b 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -391,6 +391,193 @@
}
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session)
+{
+ ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("startOutput() unknow output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1) {
+ audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
+ uint32_t waitMs = 0;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != newDevice) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate.
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+ uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams[stream].getVolumeIndex(newDevice),
+ output,
+ newDevice);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+ if (waitMs > muteWaitMs) {
+ usleep((waitMs - muteWaitMs) * 2 * 1000);
+ }
+ }
+#ifdef DOLBY_UDC
+ // It is observed that in some use-cases where both outputs are present eg. bluetooth and headphone,
+ // the output for particular stream type is decided in this routine. Hence we must call
+ // getDeviceForStrategy in order to get the current active output for this stream type and update
+ // the dolby system property.
+ if (stream == AudioSystem::MUSIC)
+ {
+ audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AudioSystem::MUSIC), true);
+ DolbySystemProperty::set(audioOutputDevice);
+ }
+#endif // DOLBY_END
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session)
+{
+ ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("stopOutput() unknow output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, false, false);
+ }
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0) {
+ outputDesc->mStopTime[stream] = systemTime();
+ audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (curOutput != output &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevice != desc->device())) {
+ setOutputDevice(curOutput,
+ getNewDevice(curOutput, false /*fromCache*/),
+ true,
+ outputDesc->mLatency*2);
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
+ }
+}
+
+audio_devices_t AudioPolicyManager::getNewDevice(audio_io_handle_t output, bool fromCache)
+{
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ AudioOutputDescriptor *primaryOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 4: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 5: the strategy media is active on the output:
+ // use device for strategy media
+ // 6: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)||
+ (primaryOutputDesc->isStrategyActive(STRATEGY_SONIFICATION)&& !primaryOutputDesc->isStrategyActive(STRATEGY_MEDIA))){
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ }
+
+ ALOGV("getNewDevice() selected device %x", device);
+ return device;
+}
+
+//private function, no changes from AudioPolicyManagerBase
+void AudioPolicyManager::handleNotificationRoutingForStream(AudioSystem::stream_type stream) {
+ switch(stream) {
+ case AudioSystem::MUSIC:
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ updateDevicesAndOutputs();
+ break;
+ default:
+ break;
+ }
+}
+
audio_io_handle_t AudioPolicyManager::getInput(int inputSource,
uint32_t samplingRate,
uint32_t format,
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 188488a..ca7031d 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -61,6 +61,12 @@
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
virtual void setPhoneState(int state);
+ virtual status_t startOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session = 0);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session = 0);
protected:
// return the strategy corresponding to a given stream type
static routing_strategy getStrategy(AudioSystem::stream_type stream);
@@ -88,6 +94,7 @@
// check that volume change is permitted, compute and send new volume to audio hardware
status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+ audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
// returns the category the device belongs to with regard to volume curve management
static device_category getDeviceCategory(audio_devices_t device);
@@ -99,5 +106,8 @@
//parameter indicates if HDMI plug in/out detected
bool mHdmiAudioEvent;
+private:
+ void handleNotificationRoutingForStream(AudioSystem::stream_type stream);
+
};
};
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index 9e6c1fc..657195d 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -153,16 +153,27 @@
bass_ctxt->device = device;
if((device == AUDIO_DEVICE_OUT_SPEAKER) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) ||
- (device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER)) {
+ (device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER) ||
+ (device == AUDIO_DEVICE_OUT_PROXY) ||
+ (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+ (device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET)) {
if (offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass))) {
offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), false);
bass_ctxt->temp_disabled = true;
+ if (bass_ctxt->ctl)
+ offload_bassboost_send_params(bass_ctxt->ctl,
+ bass_ctxt->offload_bass,
+ OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG);
}
} else {
if (!offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass)) &&
bass_ctxt->temp_disabled) {
offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), true);
bass_ctxt->temp_disabled = false;
+ if (bass_ctxt->ctl)
+ offload_bassboost_send_params(bass_ctxt->ctl,
+ bass_ctxt->offload_bass,
+ OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG);
}
}
offload_bassboost_set_device(&(bass_ctxt->offload_bass), device);
@@ -211,7 +222,9 @@
bassboost_context_t *bass_ctxt = (bassboost_context_t *)context;
ALOGV("%s", __func__);
- if (!offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass))) {
+
+ if (!offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass)) &&
+ !(bass_ctxt->temp_disabled)) {
offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), true);
if (bass_ctxt->ctl && bass_ctxt->strength)
offload_bassboost_send_params(bass_ctxt->ctl,
diff --git a/post_proc/effect_api.c b/post_proc/effect_api.c
index 0b77969..b7cf469 100644
--- a/post_proc/effect_api.c
+++ b/post_proc/effect_api.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, 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
@@ -391,7 +391,7 @@
{
ALOGV("%s", __func__);
if (preset && (preset <= NUM_OSL_REVERB_PRESETS_SUPPORTED))
- reverb->preset = map_reverb_opensl_preset_2_offload_preset[preset][1];
+ reverb->preset = map_reverb_opensl_preset_2_offload_preset[preset-1][1];
}
void offload_reverb_set_wet_mix(struct reverb_params *reverb, int wet_mix)
diff --git a/post_proc/reverb.c b/post_proc/reverb.c
index 4c7fe25..7c50430 100644
--- a/post_proc/reverb.c
+++ b/post_proc/reverb.c
@@ -309,6 +309,7 @@
return -EINVAL;
*(uint16_t *)value = reverb_ctxt->next_preset;
ALOGV("get REVERB_PARAM_PRESET, preset %d", reverb_ctxt->next_preset);
+ return 0;
}
switch (param) {
case REVERB_PARAM_ROOM_LEVEL:
@@ -464,6 +465,7 @@
return -EINVAL;
}
reverb_set_preset(reverb_ctxt, preset);
+ return 0;
}
switch (param) {
case REVERB_PARAM_PROPERTIES:
@@ -603,6 +605,14 @@
ALOGV("%s", __func__);
reverb_ctxt->ctl = output->ctl;
+ if (offload_reverb_get_enable_flag(&(reverb_ctxt->offload_reverb))) {
+ if (reverb_ctxt->ctl && reverb_ctxt->preset) {
+ offload_reverb_send_params(reverb_ctxt->ctl, reverb_ctxt->offload_reverb,
+ OFFLOAD_SEND_REVERB_ENABLE_FLAG |
+ OFFLOAD_SEND_REVERB_PRESET);
+ }
+ }
+
return 0;
}
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 4190129..54bca07 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -153,16 +153,27 @@
virt_ctxt->device = device;
if((device == AUDIO_DEVICE_OUT_SPEAKER) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) ||
- (device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER)) {
+ (device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER) ||
+ (device == AUDIO_DEVICE_OUT_PROXY) ||
+ (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+ (device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET)) {
if (offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt))) {
offload_virtualizer_set_enable_flag(&(virt_ctxt->offload_virt), false);
virt_ctxt->temp_disabled = true;
+ if (virt_ctxt->ctl)
+ offload_virtualizer_send_params(virt_ctxt->ctl,
+ virt_ctxt->offload_virt,
+ OFFLOAD_SEND_VIRTUALIZER_ENABLE_FLAG);
}
} else {
if (!offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt)) &&
virt_ctxt->temp_disabled) {
offload_virtualizer_set_enable_flag(&(virt_ctxt->offload_virt), true);
virt_ctxt->temp_disabled = false;
+ if (virt_ctxt->ctl)
+ offload_virtualizer_send_params(virt_ctxt->ctl,
+ virt_ctxt->offload_virt,
+ OFFLOAD_SEND_VIRTUALIZER_ENABLE_FLAG);
}
}
offload_virtualizer_set_device(&(virt_ctxt->offload_virt), device);
@@ -210,7 +221,9 @@
virtualizer_context_t *virt_ctxt = (virtualizer_context_t *)context;
ALOGV("%s", __func__);
- if (!offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt))) {
+
+ if (!offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt)) &&
+ !(virt_ctxt->temp_disabled)) {
offload_virtualizer_set_enable_flag(&(virt_ctxt->offload_virt), true);
if (virt_ctxt->ctl && virt_ctxt->strength)
offload_virtualizer_send_params(virt_ctxt->ctl,