hal: fix compiler warnings which are now treated as errors
With CLANG enabled and -Wall & -Werror being set in
LOCAL_CFLAGS, all warnings are treated as errors.
This commit fixes all the warnings most of which are
about unused variables and functions.
Change-Id: I32c6593fa0ad6a4fdca5dc8e6c76982a146a8bc6
diff --git a/hal/Android.mk b/hal/Android.mk
index 4c06288..c1e122d 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -285,7 +285,7 @@
LOCAL_SHARED_LIBRARIES += libperipheral_client
endif
-#LOCAL_CFLAGS += -Wall -Werror
+LOCAL_CFLAGS += -Wall -Werror
LOCAL_COPY_HEADERS_TO := mm-audio
LOCAL_COPY_HEADERS := audio_extn/audio_defs.h
diff --git a/hal/audio_extn/fm.c b/hal/audio_extn/fm.c
index 54700ba..a28d52f 100644
--- a/hal/audio_extn/fm.c
+++ b/hal/audio_extn/fm.c
@@ -114,7 +114,7 @@
static int32_t fm_stop(struct audio_device *adev)
{
- int32_t i, ret = 0;
+ int32_t ret = 0;
struct audio_usecase *uc_info;
ALOGD("%s: enter", __func__);
@@ -153,7 +153,7 @@
static int32_t fm_start(struct audio_device *adev)
{
- int32_t i, ret = 0;
+ int32_t ret = 0;
struct audio_usecase *uc_info;
int32_t pcm_dev_rx_id, pcm_dev_tx_id;
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 5a5afcc..1fba5c6 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -133,7 +133,7 @@
static int32_t start_hfp(struct audio_device *adev,
struct str_parms *parms __unused)
{
- int32_t i, ret = 0;
+ int32_t ret = 0;
struct audio_usecase *uc_info;
int32_t pcm_dev_rx_id, pcm_dev_tx_id, pcm_dev_asm_rx_id, pcm_dev_asm_tx_id;
@@ -243,7 +243,7 @@
static int32_t stop_hfp(struct audio_device *adev)
{
- int32_t i, ret = 0;
+ int32_t ret = 0;
struct audio_usecase *uc_info;
ALOGD("%s: enter", __func__);
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 8882b90..7e37efc 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -170,7 +170,6 @@
void audio_extn_sound_trigger_stop_lab(struct stream_in *in)
{
- int status = 0;
struct sound_trigger_info *st_ses_info = NULL;
audio_event_info_t event;
@@ -341,7 +340,6 @@
{
int status = 0;
char sound_trigger_lib[100];
- void *lib_handle;
ALOGI("%s: Enter", __func__);
diff --git a/hal/audio_extn/source_track.c b/hal/audio_extn/source_track.c
index 8bf4c67..bbe876b 100644
--- a/hal/audio_extn/source_track.c
+++ b/hal/audio_extn/source_track.c
@@ -255,9 +255,8 @@
static int parse_soundfocus_sourcetracking_keys(struct str_parms *parms)
{
- char *str;
char *value = NULL;
- int val, len;
+ int len;
int ret = 0, err;
char *kv_pairs = str_parms_to_str(parms);
@@ -339,7 +338,7 @@
char sound_focus_mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = "Sound Focus";
char source_tracking_mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = "Source Tracking";
int ret = -EINVAL;
- int i, count;
+ int count;
if (bitmask & BITMASK_AUDIO_PARAMETER_KEYS_SOUND_FOCUS) {
/* Derive the mixer control name based on the use case and the audio interface
@@ -434,7 +433,7 @@
const struct sound_focus_param sound_focus_data,
const struct source_tracking_param source_tracking_data)
{
- int i = 0, len = 0;
+ int i = 0;
char value[MAX_STR_SIZE] = "";
if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES) {
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 44ff952..1a616df 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -242,7 +242,6 @@
static void spkr_prot_set_spkrstatus(bool enable)
{
- struct timespec ts;
if (enable)
handle.spkr_in_use = true;
else {
@@ -255,7 +254,6 @@
{
pthread_t threadid;
struct audio_usecase *uc_info;
- int count = 0;
threadid = pthread_self();
ALOGV("%s: Entry", __func__);
if (pthread_equal(handle.speaker_prot_threadid, threadid) || !adev) {
@@ -657,7 +655,6 @@
{
unsigned long sec = 0;
int t0;
- int i = 0;
int t0_spk_1 = 0;
int t0_spk_2 = 0;
bool goahead = false;
@@ -1131,9 +1128,8 @@
int audio_extn_fbsp_set_parameters(struct str_parms *parms)
{
int ret= 0 , err;
- char *str;
char *value = NULL;
- int val, len, i;
+ int len;
char *test_r = NULL;
char *cfg_str;
int wait_time, ftm_time;
@@ -1241,7 +1237,6 @@
get_spkr_prot_ftm_param(value);
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_FBSP_GET_FTM_PARAM, value);
}
-done:
return err;
}
diff --git a/hal/audio_extn/ssr.c b/hal/audio_extn/ssr.c
index cf393c5..6534385 100644
--- a/hal/audio_extn/ssr.c
+++ b/hal/audio_extn/ssr.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -155,10 +155,6 @@
static void *ssr_process_thread(void *context);
-/* Use AAC/DTS channel mapping as default channel mapping: C,FL,FR,Ls,Rs,LFE */
-static const int chan_map[] = { 1, 2, 4, 3, 0, 5};
-
-
static int32_t drc_init_lib(int num_chan, int sample_rate __unused)
{
int ret = 0;
@@ -424,7 +420,6 @@
{
uint32_t ret = -1;
char c_multi_ch_dump[128] = {0};
- char c_ssr_3mic[128] = {0};
uint32_t buffer_size;
ALOGD("%s: ssr case, sample rate %d", __func__, in->config.rate);
@@ -559,8 +554,6 @@
int32_t audio_extn_ssr_deinit()
{
- int i;
-
ALOGV("%s: entry", __func__);
deinit_ssr_process_thread();
@@ -681,7 +674,6 @@
void *buffer, size_t bytes)
{
struct stream_in *in = (struct stream_in *)stream;
- struct audio_device *adev = in->dev;
int32_t ret = 0;
struct pcm_buffer_queue *in_buf;
struct pcm_buffer_queue *out_buf;
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 8f49b8d..846f88f 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -391,7 +391,6 @@
void audio_extn_utils_dump_streams_output_cfg_list(
struct listnode *streams_output_cfg_list)
{
- int i=0;
struct listnode *node_i, *node_j;
struct streams_output_cfg *so_info;
struct stream_format *sf_info;
@@ -418,7 +417,6 @@
{
struct listnode *node_i, *node_j;
struct streams_output_cfg *so_info;
- struct stream_format *sf_info;
ALOGV("%s", __func__);
while (!list_empty(streams_output_cfg_list)) {
@@ -490,10 +488,9 @@
audio_channel_mask_t channel_mask,
struct stream_app_type_cfg *app_type_cfg)
{
- struct listnode *node_i, *node_j, *node_k;
+ struct listnode *node_i, *node_j;
struct streams_output_cfg *so_info;
struct stream_format *sf_info;
- struct stream_sample_rate *ss_info;
char value[PROPERTY_VALUE_MAX] = {0};
if ((24 == bit_width) &&
@@ -567,7 +564,7 @@
char mixer_ctl_name[MAX_LENGTH_MIXER_CONTROL_IN_INT];
int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc;
struct mixer_ctl *ctl;
- int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
+ int pcm_device_id = 0, acdb_dev_id, snd_device = usecase->out_snd_device;
int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
char value[PROPERTY_VALUE_MAX] = {0};
@@ -855,7 +852,6 @@
default:
break;
}
-done:
outp[k] = '\0';
return k;
}
@@ -890,8 +886,6 @@
void get_default_compressed_channel_status(
unsigned char *channel_status)
{
- int32_t status = 0;
- unsigned char bit_index;
memset(channel_status,0,24);
/* block start bit in preamble bit 3 */
@@ -937,7 +931,6 @@
unsigned char *channel_status)
{
int32_t status = 0;
- unsigned char bit_index;
memset(channel_status,0,24);
/* block start bit in preamble bit 3 */
channel_status[0] |= PROFESSIONAL;
@@ -989,7 +982,7 @@
struct snd_aes_iec958 iec958;
const char *mixer_ctl_name = "IEC958 Playback PCM Stream";
struct mixer_ctl *ctl;
- int i=0;
+ ALOGV("%s: buffer %s bytes %zd", __func__, buffer, bytes);
#ifdef HDMI_PASSTHROUGH_ENABLED
if (audio_extn_is_dolby_format(out->format) &&
/*TODO:Extend code to support DTS passthrough*/
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
old mode 100755
new mode 100644
index d7a3169..4840bf1
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -249,8 +249,6 @@
static pthread_mutex_t adev_init_lock;
static unsigned int audio_device_ref_count;
-static int set_voice_volume_l(struct audio_device *adev, float volume);
-
__attribute__ ((visibility ("default")))
bool audio_hw_send_gain_dep_calibration(int level) {
bool ret_val = false;
@@ -938,7 +936,6 @@
struct audio_usecase *voip_usecase = NULL;
struct audio_usecase *hfp_usecase = NULL;
audio_usecase_t hfp_ucid;
- struct listnode *node;
int status = 0;
ALOGD("%s for use case (%s)", __func__, use_case_table[uc_id]);
@@ -1130,7 +1127,7 @@
static int stop_input_stream(struct stream_in *in)
{
- int i, ret = 0;
+ int ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
@@ -1617,7 +1614,7 @@
static int stop_output_stream(struct stream_out *out)
{
- int i, ret = 0;
+ int ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
@@ -2048,12 +2045,9 @@
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
- struct audio_usecase *usecase;
- struct listnode *node;
struct str_parms *parms;
char value[32];
int ret = 0, val = 0, err;
- bool select_new_device = false;
ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
__func__, out->usecase, use_case_table[out->usecase], kvpairs);
@@ -2372,7 +2366,7 @@
ret = compress_write(out->compr, buffer, bytes);
if (ret < 0)
ret = -errno;
- ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
+ ALOGVV("%s: writing buffer (%zu bytes) to compress device returned %zd", __func__, bytes, ret);
if (ret >= 0 && ret < (ssize_t)bytes) {
ALOGD("No space available in compress driver, post msg to cb thread");
send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
@@ -2400,7 +2394,7 @@
if (out->muted)
memset((void *)buffer, 0, bytes);
- ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
+ ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes);
if (adev->adm_request_focus)
adev->adm_request_focus(adev->adm_data, out->handle);
@@ -2761,7 +2755,6 @@
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
struct str_parms *parms;
- char *str;
char value[32];
int ret = 0, val = 0, err;
@@ -2805,7 +2798,6 @@
}
}
-done:
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
@@ -2821,7 +2813,6 @@
struct stream_in *in = (struct stream_in *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str;
- char value[256];
struct str_parms *reply = str_parms_create();
if (!query || !reply) {
@@ -2858,7 +2849,7 @@
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
- int i, ret = -1;
+ int ret = -1;
int snd_scard_state = get_snd_card_state(adev);
lock_input_stream(in);
@@ -3005,7 +2996,7 @@
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
- int i, ret = 0;
+ int ret = 0;
audio_format_t format;
*stream_out = NULL;
@@ -3226,7 +3217,7 @@
//Decide if we need to use gapless mode by default
check_and_set_gapless_mode(adev);
} else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
- ret = voice_check_and_set_incall_music_usecase(adev, out);
+ ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
if (ret != 0) {
ALOGE("%s: Incall music delivery usecase cannot be set error:%d",
__func__, ret);
@@ -3416,7 +3407,6 @@
{
struct audio_device *adev = (struct audio_device *)dev;
struct str_parms *parms;
- char *str;
char value[32];
int val;
int ret;
@@ -3431,8 +3421,6 @@
if (ret >= 0) {
char *snd_card_status = value+2;
if (strstr(snd_card_status, "OFFLINE")) {
- struct listnode *node;
- struct audio_usecase *usecase;
ALOGD("Received sound card OFFLINE status");
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
//close compress sessions on OFFLINE status
@@ -3899,8 +3887,6 @@
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device)
{
- int i, ret;
-
ALOGD("%s: enter", __func__);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index fb8d648..e18f2ec 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -53,78 +53,6 @@
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
#define WSA_MIXER_PATH_EXTENSION "wsa-"
-static const snd_device_t taiko_fluid_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
-};
-
-static const snd_device_t taiko_CDP_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
- SND_DEVICE_IN_QUAD_MIC,
-};
-
-static const snd_device_t taiko_apq8084_CDP_variant_devices[] = {
- SND_DEVICE_IN_HANDSET_MIC,
-};
-
-static const snd_device_t taiko_liquid_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
- SND_DEVICE_IN_SPEAKER_MIC,
- SND_DEVICE_IN_HEADSET_MIC,
- SND_DEVICE_IN_VOICE_DMIC,
- SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
- SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
- SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
- SND_DEVICE_IN_QUAD_MIC,
- SND_DEVICE_IN_HANDSET_STEREO_DMIC,
- SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
-};
-
-static const snd_device_t taiko_DB_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
- SND_DEVICE_IN_SPEAKER_MIC,
- SND_DEVICE_IN_HEADSET_MIC,
- SND_DEVICE_IN_QUAD_MIC,
-};
-
-static const snd_device_t tapan_lite_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_VOICE_HEADPHONES,
- SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
- SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
-};
-
-static const snd_device_t tapan_skuf_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
- /*SND_DEVICE_OUT_SPEAKER_AND_ANC_FB_HEADSET,*/
-};
-
-static const snd_device_t tapan_lite_skuf_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_VOICE_HEADPHONES,
- SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
- SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
-};
-
-static const snd_device_t helicon_skuab_variant_devices[] = {
- SND_DEVICE_OUT_SPEAKER,
- SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
- SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
-};
-
static const snd_device_t wsa_combo_devices[] = {
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_LINE,
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 923135e..17e496d 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -1076,7 +1076,7 @@
}
void platform_set_echo_reference(struct audio_device *adev, bool enable,
- audio_devices_t out_device)
+ audio_devices_t out_device __unused)
{
struct platform_data *my_data = (struct platform_data *)adev->platform;
@@ -1121,140 +1121,6 @@
}
}
-static struct csd_data *open_csd_client()
-{
- struct csd_data *csd = calloc(1, sizeof(struct csd_data));
- if (!csd) {
- ALOGE("failed to allocate csd_data mem");
- return NULL;
- }
-
- csd->csd_client = dlopen(LIB_CSD_CLIENT, RTLD_NOW);
- if (csd->csd_client == NULL) {
- ALOGE("%s: DLOPEN failed for %s", __func__, LIB_CSD_CLIENT);
- goto error;
- } else {
- ALOGV("%s: DLOPEN successful for %s", __func__, LIB_CSD_CLIENT);
-
- csd->deinit = (deinit_t)dlsym(csd->csd_client,
- "csd_client_deinit");
- if (csd->deinit == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_deinit", __func__,
- dlerror());
- goto error;
- }
- csd->disable_device = (disable_device_t)dlsym(csd->csd_client,
- "csd_client_disable_device");
- if (csd->disable_device == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_disable_device",
- __func__, dlerror());
- goto error;
- }
- csd->enable_device_config = (enable_device_config_t)dlsym(csd->csd_client,
- "csd_client_enable_device_config");
- if (csd->enable_device_config == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_enable_device_config",
- __func__, dlerror());
- goto error;
- }
- csd->enable_device = (enable_device_t)dlsym(csd->csd_client,
- "csd_client_enable_device");
- if (csd->enable_device == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_enable_device",
- __func__, dlerror());
- goto error;
- }
- csd->start_voice = (start_voice_t)dlsym(csd->csd_client,
- "csd_client_start_voice");
- if (csd->start_voice == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_start_voice",
- __func__, dlerror());
- goto error;
- }
- csd->stop_voice = (stop_voice_t)dlsym(csd->csd_client,
- "csd_client_stop_voice");
- if (csd->stop_voice == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_stop_voice",
- __func__, dlerror());
- goto error;
- }
- csd->volume = (volume_t)dlsym(csd->csd_client,
- "csd_client_volume");
- if (csd->volume == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_volume",
- __func__, dlerror());
- goto error;
- }
- csd->mic_mute = (mic_mute_t)dlsym(csd->csd_client,
- "csd_client_mic_mute");
- if (csd->mic_mute == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_mic_mute",
- __func__, dlerror());
- goto error;
- }
- csd->slow_talk = (slow_talk_t)dlsym(csd->csd_client,
- "csd_client_slow_talk");
- if (csd->slow_talk == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_slow_talk",
- __func__, dlerror());
- goto error;
- }
- csd->start_playback = (start_playback_t)dlsym(csd->csd_client,
- "csd_client_start_playback");
- if (csd->start_playback == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_start_playback",
- __func__, dlerror());
- goto error;
- }
- csd->stop_playback = (stop_playback_t)dlsym(csd->csd_client,
- "csd_client_stop_playback");
- if (csd->stop_playback == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_stop_playback",
- __func__, dlerror());
- goto error;
- }
- csd->set_lch = (set_lch_t)dlsym(csd->csd_client, "csd_client_set_lch");
- if (csd->set_lch == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_set_lch",
- __func__, dlerror());
- /* Ignore the error as this is not mandatory function for
- * basic voice call to work.
- */
- }
- csd->start_record = (start_record_t)dlsym(csd->csd_client,
- "csd_client_start_record");
- if (csd->start_record == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_start_record",
- __func__, dlerror());
- goto error;
- }
- csd->stop_record = (stop_record_t)dlsym(csd->csd_client,
- "csd_client_stop_record");
- if (csd->stop_record == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_stop_record",
- __func__, dlerror());
- goto error;
- }
-
-
- csd->init = (init_t)dlsym(csd->csd_client, "csd_client_init");
-
- if (csd->init == NULL) {
- ALOGE("%s: dlsym error %s for csd_client_init",
- __func__, dlerror());
- goto error;
- } else {
- csd->init();
- }
- }
- return csd;
-
-error:
- free(csd);
- csd = NULL;
- return csd;
-}
-
void close_csd_client(struct csd_data *csd)
{
if (csd != NULL) {
@@ -1269,7 +1135,6 @@
static void set_platform_defaults()
{
int32_t dev, count = 0;
- char dsp_decoder_property[PROPERTY_VALUE_MAX];
const char *MEDIA_MIMETYPE_AUDIO_ALAC = "audio/alac";
const char *MEDIA_MIMETYPE_AUDIO_APE = "audio/x-ape";
@@ -1636,14 +1501,11 @@
void *platform_init(struct audio_device *adev)
{
- char platform[PROPERTY_VALUE_MAX];
- char baseband[PROPERTY_VALUE_MAX];
char value[PROPERTY_VALUE_MAX];
struct platform_data *my_data = NULL;
- int retry_num = 0, snd_card_num = 0, key = 0;
+ int retry_num = 0, snd_card_num = 0;
const char *snd_card_name;
char mixer_xml_path[MAX_MIXER_XML_PATH],ffspEnable[PROPERTY_VALUE_MAX];
- char *cvd_version = NULL;
const char *mixer_ctl_name = "Set HPX ActiveBe";
struct mixer_ctl *ctl = NULL;
int idx;
@@ -2213,7 +2075,8 @@
return DEFAULT_APP_TYPE;
}
-int platform_get_default_app_type_v2(void *platform, usecase_type_t type)
+int platform_get_default_app_type_v2(void *platform __unused,
+ usecase_type_t type __unused)
{
if(type == PCM_CAPTURE)
return DEFAULT_APP_TYPE_TX_PATH;
@@ -2413,7 +2276,6 @@
{
struct platform_data *my_data = (struct platform_data *)platform;
int acdb_dev_id, acdb_dev_type;
- struct audio_device *adev = my_data->adev;
int snd_device = SND_DEVICE_OUT_SPEAKER;
if (usecase->type == PCM_PLAYBACK)
@@ -2595,7 +2457,8 @@
return ret;
}
-int platform_get_sample_rate(void *platform, uint32_t *rate)
+int platform_get_sample_rate(void *platform __unused,
+ uint32_t *rate __unused)
{
return 0;
}
@@ -3297,7 +3160,6 @@
int max_channels = 2;
int i = 0, ret = 0;
struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
edid_audio_info *info = NULL;
ret = platform_get_edid_info(platform);
info = (edid_audio_info *)my_data->edid_info;
@@ -3374,168 +3236,11 @@
return ret;
}
-static int update_external_device_status(struct platform_data *my_data,
- char* event_name, bool status)
-{
- int ret = 0;
- struct audio_usecase *usecase;
- struct listnode *node;
-
- ALOGD("Recieved external event switch %s", event_name);
-
- if (!strcmp(event_name, EVENT_EXTERNAL_SPK_1))
- my_data->external_spk_1 = status;
- else if (!strcmp(event_name, EVENT_EXTERNAL_SPK_2))
- my_data->external_spk_2 = status;
- else if (!strcmp(event_name, EVENT_EXTERNAL_MIC))
- my_data->external_mic = status;
- else {
- ALOGE("The audio event type is not found");
- return -EINVAL;
- }
-
- list_for_each(node, &my_data->adev->usecase_list) {
- usecase = node_to_item(node, struct audio_usecase, list);
- select_devices(my_data->adev, usecase->id);
- }
-
- return ret;
-}
-
-static int parse_audiocal_cfg(struct str_parms *parms, acdb_audio_cal_cfg_t *cal)
-{
- int err;
- unsigned int val;
- char value[64];
- int ret = 0;
-
- if(parms == NULL || cal == NULL)
- return ret;
-
- err = str_parms_get_str(parms, "cal_persist", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_persist");
- cal->persist = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x1;
- }
- err = str_parms_get_str(parms, "cal_apptype", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_apptype");
- cal->app_type = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x2;
- }
- err = str_parms_get_str(parms, "cal_caltype", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_caltype");
- cal->cal_type = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x4;
- }
- err = str_parms_get_str(parms, "cal_samplerate", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_samplerate");
- cal->sampling_rate = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x8;
- }
- err = str_parms_get_str(parms, "cal_devid", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_devid");
- cal->dev_id = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x10;
- }
- err = str_parms_get_str(parms, "cal_snddevid", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_snddevid");
- cal->snd_dev_id = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x20;
- }
- err = str_parms_get_str(parms, "cal_topoid", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_topoid");
- cal->topo_id = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x40;
- }
- err = str_parms_get_str(parms, "cal_moduleid", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_moduleid");
- cal->module_id = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x80;
- }
- err = str_parms_get_str(parms, "cal_paramid", value, sizeof(value));
- if (err >= 0) {
- str_parms_del(parms, "cal_paramid");
- cal->param_id = (uint32_t) strtoul(value, NULL, 0);
- ret = ret | 0x100;
- }
- return ret;
-}
-
-static void set_audiocal(void *platform, struct str_parms *parms, char *value, int len) {
- struct platform_data *my_data = (struct platform_data *)platform;
- struct stream_out out;
- acdb_audio_cal_cfg_t cal={0};
- uint8_t *dptr = NULL;
- int32_t dlen;
- int err, ret;
- if(value == NULL || platform == NULL || parms == NULL) {
- ALOGE("[%s] received null pointer, failed",__func__);
- goto done_key_audcal;
- }
-
- /* parse audio calibration keys */
- ret = parse_audiocal_cfg(parms, &cal);
-
- /* handle audio calibration data now */
- err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_AUD_CALDATA, value, len);
- if (err >= 0) {
- str_parms_del(parms, AUDIO_PARAMETER_KEY_AUD_CALDATA);
- dlen = strlen(value);
- if(dlen <= 0) {
- ALOGE("[%s] null data received",__func__);
- goto done_key_audcal;
- }
- dptr = (uint8_t*) calloc(dlen, sizeof(uint8_t));
- if(dptr == NULL) {
- ALOGE("[%s] memory allocation failed for %d",__func__, dlen);
- goto done_key_audcal;
- }
- dlen = b64decode(value, strlen(value), dptr);
- if(dlen<=0) {
- ALOGE("[%s] data decoding failed %d", __func__, dlen);
- goto done_key_audcal;
- }
-
- if(cal.dev_id) {
- if(audio_is_input_device(cal.dev_id)) {
- cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
- } else {
- out.devices = cal.dev_id;
- out.sample_rate = cal.sampling_rate;
- cal.snd_dev_id = platform_get_output_snd_device(platform, &out);
- }
- }
- cal.acdb_dev_id = platform_get_snd_device_acdb_id(cal.snd_dev_id);
- ALOGD("Setting audio calibration for snd_device(%d) acdb_id(%d)",
- cal.snd_dev_id, cal.acdb_dev_id);
- if(cal.acdb_dev_id == -EINVAL) {
- ALOGE("[%s] Invalid acdb_device id %d for snd device id %d",
- __func__, cal.acdb_dev_id, cal.snd_dev_id);
- goto done_key_audcal;
- }
- if(my_data->acdb_set_audio_cal) {
- ret = my_data->acdb_set_audio_cal((void *)&cal, (void*)dptr, dlen);
- }
- }
-done_key_audcal:
- if(dptr != NULL)
- free(dptr);
-}
-
int platform_set_parameters(void *platform, struct str_parms *parms)
{
struct platform_data *my_data = (struct platform_data *)platform;
- char *str;
char value[256] = {0};
- int val,len;
+ int len;
int ret = 0, err;
char *kv_pairs = NULL;
@@ -3710,108 +3415,11 @@
return ret;
}
-static void get_audiocal(void *platform, void *keys, void *pReply) {
- struct platform_data *my_data = (struct platform_data *)platform;
- struct stream_out out;
- struct str_parms *query = (struct str_parms *)keys;
- struct str_parms *reply=(struct str_parms *)pReply;
- acdb_audio_cal_cfg_t cal={0};
- uint8_t *dptr = NULL;
- char value[512] = {0};
- char *rparms=NULL;
- int ret=0, err;
- uint32_t param_len;
-
- if(query==NULL || platform==NULL || reply==NULL) {
- ALOGE("[%s] received null pointer",__func__);
- ret=-EINVAL;
- goto done;
- }
- /* parse audiocal configuration keys */
- ret = parse_audiocal_cfg(query, &cal);
- if(ret == 0) {
- /* No calibration keys found */
- goto done;
- }
- err = str_parms_get_str(query, AUDIO_PARAMETER_KEY_AUD_CALDATA, value, sizeof(value));
- if (err >= 0) {
- str_parms_del(query, AUDIO_PARAMETER_KEY_AUD_CALDATA);
- } else {
- goto done;
- }
-
- if(cal.dev_id & AUDIO_DEVICE_BIT_IN) {
- cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
- } else if(cal.dev_id) {
- out.devices = cal.dev_id;
- out.sample_rate = cal.sampling_rate;
- cal.snd_dev_id = platform_get_output_snd_device(platform, &out);
- }
- cal.acdb_dev_id = platform_get_snd_device_acdb_id(cal.snd_dev_id);
- if (cal.acdb_dev_id < 0) {
- ALOGE("%s: Failed. Could not find acdb id for snd device(%d)",
- __func__, cal.snd_dev_id);
- ret = -EINVAL;
- goto done_key_audcal;
- }
- ALOGD("[%s] Getting audio calibration for snd_device(%d) acdb_id(%d)",
- __func__, cal.snd_dev_id, cal.acdb_dev_id);
-
- param_len = MAX_SET_CAL_BYTE_SIZE;
- dptr = (uint8_t*)calloc(param_len, sizeof(uint8_t));
- if(dptr == NULL) {
- ALOGE("[%s] Memory allocation failed for length %d",__func__,param_len);
- ret = -ENOMEM;
- goto done_key_audcal;
- }
- if (my_data->acdb_get_audio_cal != NULL) {
- ret = my_data->acdb_get_audio_cal((void*)&cal, (void*)dptr, ¶m_len);
- if (ret == 0) {
- int dlen;
- if(param_len == 0 || param_len == MAX_SET_CAL_BYTE_SIZE) {
- ret = -EINVAL;
- goto done_key_audcal;
- }
- /* Allocate memory for encoding */
- rparms = (char*)calloc((param_len*2), sizeof(char));
- if(rparms == NULL) {
- ALOGE("[%s] Memory allocation failed for size %d",
- __func__, param_len*2);
- ret = -ENOMEM;
- goto done_key_audcal;
- }
- if(cal.persist==0 && cal.module_id && cal.param_id) {
- err = b64encode(dptr+12, param_len-12, rparms);
- } else {
- err = b64encode(dptr, param_len, rparms);
- }
- if(err < 0) {
- ALOGE("[%s] failed to convert data to string", __func__);
- ret = -EINVAL;
- goto done_key_audcal;
- }
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_AUD_CALRESULT, ret);
- str_parms_add_str(reply, AUDIO_PARAMETER_KEY_AUD_CALDATA, rparms);
- }
- }
-done_key_audcal:
- if(ret != 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_AUD_CALRESULT, ret);
- str_parms_add_str(reply, AUDIO_PARAMETER_KEY_AUD_CALDATA, "");
- }
-done:
- if(dptr != NULL)
- free(dptr);
- if(rparms != NULL)
- free(rparms);
-}
-
void platform_get_parameters(void *platform,
struct str_parms *query,
struct str_parms *reply)
{
struct platform_data *my_data = (struct platform_data *)platform;
- char *str = NULL;
char value[512] = {0};
int ret;
char *kv_pairs = NULL;
@@ -4119,9 +3727,6 @@
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- const char *snd_card_name = mixer_get_name(adev->mixer);
- int na_mode = platform_get_native_support();
-
backend_idx = platform_get_backend_index(snd_device);
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
@@ -4383,7 +3988,6 @@
unsigned int new_bit_width;
unsigned int new_sample_rate;
int backend_idx = DEFAULT_CODEC_BACKEND;
- struct platform_data *my_data = (struct platform_data *)adev->platform;
backend_idx = platform_get_backend_index(snd_device);
@@ -4544,10 +4148,7 @@
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
- char *sad = block;
- int num_audio_blocks;
- int channel_count = 2;
- int i, ret, count;
+ int ret, count;
struct mixer_ctl *ctl;
char edid_data[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE + 1] = {0};
@@ -4635,7 +4236,6 @@
int ret;
unsigned int i;
int set_values[8] = {0};
- char device_num[13]; // device number up to 2 digit
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
ALOGV("%s channel_count:%d",__func__, ch_count);
@@ -4733,10 +4333,8 @@
bool platform_is_edid_supported_format(void *platform, int format)
{
struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
edid_audio_info *info = NULL;
- int num_audio_blocks;
- int i, ret, count;
+ int i, ret;
unsigned char format_id = platform_map_to_edid_format(format);
ret = platform_get_edid_info(platform);
@@ -4763,11 +4361,9 @@
int platform_set_edid_channels_configuration(void *platform, int channels) {
struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
edid_audio_info *info = NULL;
- int num_audio_blocks;
int channel_count = 2;
- int i, ret, count;
+ int i, ret;
char default_channelMap[MAX_CHANNELS_SUPPORTED] = {0};
ret = platform_get_edid_info(platform);
@@ -5046,7 +4642,6 @@
const char *spkr_1_tz_name, const char *spkr_2_tz_name)
{
int ret = 0;
- int i;
if (spkr_1_tz_name == NULL && spkr_2_tz_name == NULL) {
ALOGE("%s: Invalid input", __func__);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index c8edd3e..785a202 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -828,7 +828,7 @@
}
void platform_set_echo_reference(struct audio_device *adev, bool enable,
- audio_devices_t out_device)
+ audio_devices_t out_device __unused)
{
struct platform_data *my_data = (struct platform_data *)adev->platform;
@@ -1029,8 +1029,8 @@
static void set_platform_defaults()
{
- int32_t dev, count = 0;
- char dsp_decoder_property[PROPERTY_VALUE_MAX];
+ int32_t dev;
+ unsigned int count = 0;
const char *MEDIA_MIMETYPE_AUDIO_ALAC = "audio/alac";
const char *MEDIA_MIMETYPE_AUDIO_APE = "audio/x-ape";
@@ -1299,9 +1299,8 @@
char baseband[PROPERTY_VALUE_MAX];
char value[PROPERTY_VALUE_MAX];
struct platform_data *my_data = NULL;
- int retry_num = 0, snd_card_num = 0, key = 0;
+ int retry_num = 0, snd_card_num = 0;
char *snd_card_name = NULL, *snd_card_name_t = NULL;
- char *cvd_version = NULL;
char *snd_internal_name = NULL;
char *tmp = NULL;
char mixer_xml_file[MIXER_PATH_MAX_LENGTH]= {0};
@@ -2092,7 +2091,6 @@
{
struct platform_data *my_data = (struct platform_data *)platform;
int acdb_dev_id, acdb_dev_type;
- struct audio_device *adev = my_data->adev;
int snd_device = SND_DEVICE_OUT_SPEAKER;
if (usecase->type == PCM_PLAYBACK)
@@ -2920,7 +2918,6 @@
int max_channels = 2;
int i = 0, ret = 0;
struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
edid_audio_info *info = NULL;
ret = platform_get_edid_info(platform);
info = (edid_audio_info *)my_data->edid_info;
@@ -3027,7 +3024,6 @@
static int parse_audiocal_cfg(struct str_parms *parms, acdb_audio_cal_cfg_t *cal)
{
int err;
- unsigned int val;
char value[64];
int ret = 0;
@@ -3094,7 +3090,7 @@
static void set_audiocal(void *platform, struct str_parms *parms, char *value, int len) {
struct platform_data *my_data = (struct platform_data *)platform;
struct stream_out out;
- acdb_audio_cal_cfg_t cal={0};
+ acdb_audio_cal_cfg_t cal;
uint8_t *dptr = NULL;
int32_t dlen;
int err, ret;
@@ -3193,9 +3189,8 @@
int platform_set_parameters(void *platform, struct str_parms *parms)
{
struct platform_data *my_data = (struct platform_data *)platform;
- char *str;
char *value=NULL;
- int val, len;
+ int len;
int ret = 0, err;
char *kv_pairs = str_parms_to_str(parms);
@@ -3403,7 +3398,7 @@
struct stream_out out;
struct str_parms *query = (struct str_parms *)keys;
struct str_parms *reply=(struct str_parms *)pReply;
- acdb_audio_cal_cfg_t cal={0};
+ acdb_audio_cal_cfg_t cal;
uint8_t *dptr = NULL;
char value[512] = {0};
char *rparms=NULL;
@@ -3455,7 +3450,6 @@
if (my_data->acdb_get_audio_cal != NULL) {
ret = my_data->acdb_get_audio_cal((void*)&cal, (void*)dptr, ¶m_len);
if (ret == 0) {
- int dlen;
if(param_len == 0 || param_len == MAX_SET_CAL_BYTE_SIZE) {
ret = -EINVAL;
goto done_key_audcal;
@@ -3499,7 +3493,6 @@
struct str_parms *reply)
{
struct platform_data *my_data = (struct platform_data *)platform;
- char *str = NULL;
char value[512] = {0};
int ret;
char *kv_pairs = NULL;
@@ -3566,7 +3559,6 @@
str_parms_add_int(reply, AUDIO_PARAMETER_IS_HW_DECODER_SESSION_ALLOWED, isallowed);
}
-done:
kv_pairs = str_parms_to_str(reply);
ALOGV_IF(kv_pairs != NULL, "%s: exit: returns - %s", __func__, kv_pairs);
free(kv_pairs);
@@ -3818,7 +3810,6 @@
{
bool backend_change = false;
struct listnode *node;
- struct stream_out *out = NULL;
unsigned int bit_width;
unsigned int sample_rate;
int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3946,7 +3937,6 @@
unsigned int new_bit_width;
unsigned int new_sample_rate;
int backend_idx = DEFAULT_CODEC_BACKEND;
- struct platform_data *my_data = (struct platform_data *)adev->platform;
backend_idx = platform_get_backend_index(snd_device);
@@ -4108,10 +4098,7 @@
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
- char *sad = block;
- int num_audio_blocks;
- int channel_count = 2;
- int i, ret, count;
+ int ret, count;
struct mixer_ctl *ctl;
char edid_data[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE + 1] = {0};
@@ -4200,7 +4187,6 @@
int ret;
unsigned int i;
int set_values[8] = {0};
- char device_num[13]; // device number up to 2 digit
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
ALOGV("%s channel_count:%d",__func__, ch_count);
@@ -4298,10 +4284,8 @@
bool platform_is_edid_supported_format(void *platform, int format)
{
struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
edid_audio_info *info = NULL;
- int num_audio_blocks;
- int i, ret, count;
+ int i, ret;
unsigned char format_id = platform_map_to_edid_format(format);
ret = platform_get_edid_info(platform);
@@ -4328,11 +4312,9 @@
int platform_set_edid_channels_configuration(void *platform, int channels) {
struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
edid_audio_info *info = NULL;
- int num_audio_blocks;
int channel_count = 2;
- int i, ret, count;
+ int i, ret;
char default_channelMap[MAX_CHANNELS_SUPPORTED] = {0};
ret = platform_get_edid_info(platform);
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 92cc930..2be66c4 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014-2015, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-2016, 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
@@ -306,7 +306,7 @@
}
static void process_tz_name(const XML_Char **attr)
{
- int ret, index;
+ int index;
if (strcmp(attr[0], "name") != 0) {
ALOGE("%s: 'name' not found, no Audio Interface set!", __func__);
@@ -361,10 +361,6 @@
static void start_tag(void *userdata __unused, const XML_Char *tag_name,
const XML_Char **attr)
{
- const XML_Char *attr_name = NULL;
- const XML_Char *attr_value = NULL;
- unsigned int i;
-
if (strcmp(tag_name, "bit_width_configs") == 0) {
section = BITWIDTH;
} else if (strcmp(tag_name, "acdb_ids") == 0) {
diff --git a/hal/voice.c b/hal/voice.c
index d1db987..bff69a1 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -111,7 +111,7 @@
int voice_stop_usecase(struct audio_device *adev, audio_usecase_t usecase_id)
{
- int i, ret = 0;
+ int ret = 0;
struct audio_usecase *uc_info;
struct voice_session *session = NULL;
@@ -164,7 +164,7 @@
int voice_start_usecase(struct audio_device *adev, audio_usecase_t usecase_id)
{
- int i, ret = 0;
+ int ret = 0;
struct audio_usecase *uc_info;
int pcm_dev_rx_id, pcm_dev_tx_id;
uint32_t sample_rate = 8000;
@@ -316,7 +316,6 @@
{
int ret = 0;
uint32_t session_id;
- int usecase_id;
int rec_mode = INCALL_REC_NONE;
if (voice_is_call_state_active(adev)) {
@@ -424,20 +423,6 @@
return incall_record_device;
}
-int voice_check_and_set_incall_music_usecase(struct audio_device *adev,
- struct stream_out *out)
-{
- int ret = 0;
-
- ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
- if (ret == -ENOSYS) {
- /* Incall music delivery is used only for LCH call state */
- ret = -EINVAL;
- }
-
- return ret;
-}
-
int voice_set_mic_mute(struct audio_device *adev, bool state)
{
int err = 0;
@@ -519,9 +504,7 @@
int voice_set_parameters(struct audio_device *adev, struct str_parms *parms)
{
- char *str;
char value[32];
- int val;
int ret = 0, err;
char *kv_pairs = str_parms_to_str(parms);
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 7404617..7293485 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -242,7 +242,7 @@
static int voip_stop_call(struct audio_device *adev)
{
- int i, ret = 0;
+ int ret = 0;
struct audio_usecase *uc_info;
ALOGD("%s: enter, out_stream_count=%d, in_stream_count=%d",
@@ -288,7 +288,7 @@
static int voip_start_call(struct audio_device *adev,
struct pcm_config *voip_config)
{
- int i, ret = 0;
+ int ret = 0;
struct audio_usecase *uc_info;
int pcm_dev_rx_id, pcm_dev_tx_id;
unsigned int flags = PCM_OUT | PCM_MONOTONIC;
@@ -384,10 +384,8 @@
int voice_extn_compress_voip_set_parameters(struct audio_device *adev,
struct str_parms *parms)
{
- char *str;
char value[32]={0};
int ret = 0, err, rate;
- int min_rate, max_rate;
bool flag;
char *kv_pairs = str_parms_to_str(parms);
@@ -410,7 +408,6 @@
voip_set_dtx(adev, flag);
}
-done:
ALOGV("%s: exit", __func__);
free(kv_pairs);
return ret;
@@ -421,7 +418,6 @@
{
int ret;
char value[32]={0};
- char *str = NULL;
ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_VOIP_OUT_STREAM_COUNT,
value, sizeof(value));
@@ -442,7 +438,7 @@
struct str_parms *query,
struct str_parms *reply)
{
- int ret, val;
+ int ret;
char value[32]={0};
ALOGD("%s: enter", __func__);
@@ -463,7 +459,7 @@
struct str_parms *query,
struct str_parms *reply)
{
- int ret, val;
+ int ret;
char value[32]={0};
char *kv_pairs = NULL;
@@ -546,7 +542,6 @@
int voice_extn_compress_voip_start_input_stream(struct stream_in *in)
{
int ret = 0;
- struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
int snd_card_status = get_snd_card_state(adev);
@@ -590,7 +585,7 @@
int voice_extn_compress_voip_open_output_stream(struct stream_out *out)
{
- int mode, ret;
+ int ret;
ALOGD("%s: enter", __func__);
@@ -636,9 +631,7 @@
int voice_extn_compress_voip_open_input_stream(struct stream_in *in)
{
- int sample_rate;
- int buffer_size,frame_size;
- int mode, ret;
+ int ret;
ALOGD("%s: enter", __func__);
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index f36a7a6..3cd3e78 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -164,7 +164,6 @@
audio_usecase_t usecase_id = 0;
enum voice_lch_mode lch_mode;
struct voice_session *session = NULL;
- int fd = 0;
int ret = 0;
ALOGD("%s: enter:", __func__);
@@ -442,7 +441,6 @@
int voice_extn_set_parameters(struct audio_device *adev,
struct str_parms *parms)
{
- char *str;
int value;
int ret = 0, err;
char *kv_pairs = str_parms_to_str(parms);
diff --git a/hal/voice_extn/voice_extn.h b/hal/voice_extn/voice_extn.h
index 989ee79..dcce0ab 100644
--- a/hal/voice_extn/voice_extn.h
+++ b/hal/voice_extn/voice_extn.h
@@ -105,11 +105,7 @@
int voice_extn_check_and_set_incall_music_usecase(struct audio_device *adev,
struct stream_out *out);
#else
-static int voice_extn_check_and_set_incall_music_usecase(struct audio_device *adev __unused,
- struct stream_out *out __unused)
-{
- return -ENOSYS;
-}
+#define voice_extn_check_and_set_incall_music_usecase(adev, out) -ENOSYS
#endif
#ifdef COMPRESS_VOIP_ENABLED