Merge "hal: Reject hardware NT decoder session during SSR"
diff --git a/hal/Android.mk b/hal/Android.mk
index 4e5f846..e0149f9 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -149,7 +149,7 @@
LOCAL_SRC_FILES += audio_extn/dolby.c
endif
-ifeq ($(strip $(DOLBY_DAP)),true)
+ifeq ($(strip $(DS1_DOLBY_DAP)),true)
LOCAL_CFLAGS += -DDS1_DOLBY_DAP_ENABLED
ifneq ($(strip $(DOLBY_DDP)),true)
LOCAL_SRC_FILES += audio_extn/dolby.c
@@ -205,8 +205,9 @@
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP)),true)
LOCAL_CFLAGS += -DDS2_DOLBY_DAP_ENABLED
+ LOCAL_CFLAGS += -DDS1_DOLBY_DDP_ENABLED
ifneq ($(strip $(DOLBY_DDP)),true)
- ifneq ($(strip $(DOLBY_DAP)),true)
+ ifneq ($(strip $(DS1_DOLBY_DAP)),true)
LOCAL_SRC_FILES += audio_extn/dolby.c
endif
endif
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index ca5fd59..67d30ca 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -179,6 +179,7 @@
if (adev->offload_effects_set_hpx_state != NULL)
adev->offload_effects_set_hpx_state(hpx_state);
+ audio_extn_dts_eagle_fade(adev, aextnmod.hpx_enabled, NULL);
/* set HPX state on device pp */
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (ctl)
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 98b2672..e7e2418 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -421,6 +421,7 @@
audio_format_t format,
uint32_t sample_rate,
uint32_t bit_width,
+ audio_channel_mask_t channel_mask,
struct stream_app_type_cfg *app_type_cfg);
int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase);
void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 82b596f..dbd54f7 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -134,14 +134,15 @@
static audio_output_flags_t parse_flag_names(char *name)
{
uint32_t flag = 0;
- char *flag_name = strtok(name, "|");
+ char *last_r;
+ char *flag_name = strtok_r(name, "|", &last_r);
while (flag_name != NULL) {
if (strlen(flag_name) != 0) {
flag |= string_to_enum(s_flag_name_to_enum_table,
ARRAY_SIZE(s_flag_name_to_enum_table),
flag_name);
}
- flag_name = strtok(NULL, "|");
+ flag_name = strtok_r(NULL, "|", &last_r);
}
ALOGV("parse_flag_names: flag - %d", flag);
@@ -151,7 +152,8 @@
static void parse_format_names(char *name, struct streams_output_cfg *so_info)
{
struct stream_format *sf_info = NULL;
- char *str = strtok(name, "|");
+ char *last_r;
+ char *str = strtok_r(name, "|", &last_r);
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0)
return;
@@ -169,7 +171,7 @@
sf_info->format = format;
list_add_tail(&so_info->format_list, &sf_info->list);
}
- str = strtok(NULL, "|");
+ str = strtok_r(NULL, "|", &last_r);
}
}
@@ -177,7 +179,8 @@
{
struct stream_sample_rate *ss_info = NULL;
uint32_t sample_rate = 48000;
- char *str = strtok(name, "|");
+ char *last_r;
+ char *str = strtok_r(name, "|", &last_r);
if (str != NULL && 0 == strcmp(str, DYNAMIC_VALUE_TAG))
return;
@@ -188,20 +191,22 @@
ALOGV("%s: sample_rate - %d", __func__, sample_rate);
if (0 != sample_rate) {
ss_info = (struct stream_sample_rate *)calloc(1, sizeof(struct stream_sample_rate));
- if (ss_info == NULL)
- break; /* return whatever was parsed */
-
+ if (!ss_info) {
+ ALOGE("%s: memory allocation failure", __func__);
+ return;
+ }
ss_info->sample_rate = sample_rate;
list_add_tail(&so_info->sample_rate_list, &ss_info->list);
}
- str = strtok(NULL, "|");
+ str = strtok_r(NULL, "|", &last_r);
}
}
static int parse_bit_width_names(char *name)
{
int bit_width = 16;
- char *str = strtok(name, "|");
+ char *last_r;
+ char *str = strtok_r(name, "|", &last_r);
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG))
bit_width = (int)strtol(str, (char **)NULL, 10);
@@ -213,7 +218,8 @@
static int parse_app_type_names(void *platform, char *name)
{
int app_type = platform_get_default_app_type(platform);
- char *str = strtok(name, "|");
+ char *last_r;
+ char *str = strtok_r(name, "|", &last_r);
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG))
app_type = (int)strtol(str, (char **)NULL, 10);
@@ -452,12 +458,14 @@
audio_format_t format,
uint32_t sample_rate,
uint32_t bit_width,
+ audio_channel_mask_t channel_mask,
struct stream_app_type_cfg *app_type_cfg)
{
struct listnode *node_i, *node_j, *node_k;
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) &&
(devices & AUDIO_DEVICE_OUT_SPEAKER)) {
@@ -468,6 +476,16 @@
ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
}
+ property_get("audio.playback.mch.downsample",value,"");
+ if (!strncmp("true", value, sizeof("true"))) {
+ if ((popcount(channel_mask) > 2) &&
+ (sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+ !(flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s: MCH session defaulting sample rate to %d",
+ __func__, sample_rate);
+ }
+ }
ALOGV("%s: flags: %x, format: %x sample_rate %d",
__func__, flags, format, sample_rate);
list_for_each(node_i, streams_output_cfg_list) {
@@ -509,6 +527,7 @@
struct mixer_ctl *ctl;
int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ char value[PROPERTY_VALUE_MAX] = {0};
ALOGV("%s", __func__);
@@ -558,6 +577,14 @@
sample_rate = out->app_type_cfg.sample_rate;
}
+ property_get("audio.playback.mch.downsample",value,"");
+ if (!strncmp("true", value, sizeof("true"))) {
+ if ((popcount(out->channel_mask) > 2) &&
+ (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+ !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ }
+
app_type_cfg[len++] = out->app_type_cfg.app_type;
app_type_cfg[len++] = acdb_dev_id;
if (((out->format == AUDIO_FORMAT_E_AC3) ||
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index ea4d214..435138d 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -420,7 +420,7 @@
audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY);
audio_extn_utils_send_audio_calibration(adev, usecase);
audio_extn_utils_send_app_type_cfg(usecase);
- strcpy(mixer_path, use_case_table[usecase->id]);
+ strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH);
platform_add_backend_name(mixer_path, snd_device);
ALOGV("%s: apply mixer and update path: %s", __func__, mixer_path);
audio_route_apply_and_update_path(adev->audio_route, mixer_path);
@@ -442,7 +442,7 @@
snd_device = usecase->in_snd_device;
else
snd_device = usecase->out_snd_device;
- strcpy(mixer_path, use_case_table[usecase->id]);
+ strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH);
platform_add_backend_name(mixer_path, snd_device);
ALOGV("%s: reset and update mixer path: %s", __func__, mixer_path);
audio_route_reset_and_update_path(adev->audio_route, mixer_path);
@@ -954,6 +954,7 @@
usecase->stream.out->format,
usecase->stream.out->sample_rate,
usecase->stream.out->bit_width,
+ usecase->stream.out->channel_mask,
&usecase->stream.out->app_type_cfg);
ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
}
@@ -1953,9 +1954,9 @@
for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
if (!first) {
- strcat(value, "|");
+ strlcat(value, "|", sizeof(value));
}
- strcat(value, out_channels_name_to_enum_table[j].name);
+ strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value));
first = false;
break;
}
@@ -1982,7 +1983,7 @@
for (j = 0; j < ARRAY_SIZE(out_formats_name_to_enum_table); j++) {
if (out_formats_name_to_enum_table[j].value == out->supported_formats[i]) {
if (!first) {
- strcat(value, "|");
+ strlcat(value, "|", sizeof(value));
}
strlcat(value, out_formats_name_to_enum_table[j].name, sizeof(value));
first = false;
@@ -2978,7 +2979,8 @@
audio_extn_utils_update_stream_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
devices, flags, format, out->sample_rate,
- out->bit_width, &out->app_type_cfg);
+ out->bit_width, out->channel_mask,
+ &out->app_type_cfg);
if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
(flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
/* Ensure the default output is not selected twice */
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 356d596..900a2d1 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -59,7 +59,7 @@
#define MIXER_XML_PATH_WCD9330 "/system/etc/mixer_paths_wcd9330.xml"
#define MIXER_XML_PATH_WCD9335 "/system/etc/mixer_paths_wcd9335.xml"
#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
-#define PLATFORM_INFO_XML_PATH_I2S "/system/etc/audio_platform_info_i2s.xml"
+#define PLATFORM_INFO_XML_PATH_EXTCODEC "/system/etc/audio_platform_info_extcodec.xml"
#define LIB_ACDB_LOADER "libacdbloader.so"
#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
@@ -246,22 +246,18 @@
[USECASE_AUDIO_PLAYBACK_OFFLOAD] =
{PLAYBACK_OFFLOAD_DEVICE, PLAYBACK_OFFLOAD_DEVICE},
#ifdef MULTIPLE_OFFLOAD_ENABLED
- [USECASE_AUDIO_PLAYBACK_OFFLOAD2] =
- {PLAYBACK_OFFLOAD_DEVICE2, PLAYBACK_OFFLOAD_DEVICE2},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD3] =
- {PLAYBACK_OFFLOAD_DEVICE3, PLAYBACK_OFFLOAD_DEVICE3},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD4] =
- {PLAYBACK_OFFLOAD_DEVICE4, PLAYBACK_OFFLOAD_DEVICE4},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD5] =
- {PLAYBACK_OFFLOAD_DEVICE5, PLAYBACK_OFFLOAD_DEVICE5},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD6] =
- {PLAYBACK_OFFLOAD_DEVICE6, PLAYBACK_OFFLOAD_DEVICE6},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD7] =
- {PLAYBACK_OFFLOAD_DEVICE7, PLAYBACK_OFFLOAD_DEVICE7},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD8] =
- {PLAYBACK_OFFLOAD_DEVICE8, PLAYBACK_OFFLOAD_DEVICE8},
- [USECASE_AUDIO_PLAYBACK_OFFLOAD9] =
- {PLAYBACK_OFFLOAD_DEVICE9, PLAYBACK_OFFLOAD_DEVICE9},
+ /* Below entries are initialized with invalid values
+ * Valid values should be updated from fnc platform_info_init()
+ * based on pcm ids defined in audio_platform_info.xml.
+ */
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD2] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD3] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD4] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD5] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD6] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD7] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD8] = {-1, -1},
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD9] = {-1, -1},
#endif
[USECASE_AUDIO_RECORD] = {AUDIO_RECORD_PCM_DEVICE, AUDIO_RECORD_PCM_DEVICE},
[USECASE_AUDIO_RECORD_COMPRESS] = {COMPRESS_CAPTURE_DEVICE, COMPRESS_CAPTURE_DEVICE},
@@ -1425,7 +1421,7 @@
DIR *dir;
struct dirent *dirent;
char file_name[10] = "wsa";
- strcat(CodecPeek, snd_card_name);
+ strlcat(CodecPeek, snd_card_name, sizeof(CodecPeek));
dir = opendir(CodecPeek);
if (dir != NULL) {
@@ -1450,8 +1446,11 @@
set_platform_defaults();
- /* Initialize ACDB ID's */
- platform_info_init(PLATFORM_INFO_XML_PATH);
+ /* Initialize ACDB and PCM ID's */
+ if (is_external_codec)
+ platform_info_init(PLATFORM_INFO_XML_PATH_EXTCODEC);
+ else
+ platform_info_init(PLATFORM_INFO_XML_PATH);
/* init usb */
audio_extn_usb_init(adev);
@@ -3473,6 +3472,7 @@
struct stream_out *out = NULL;
unsigned int bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
unsigned int sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ char value[PROPERTY_VALUE_MAX] = {0};
// For voice calls use default configuration
// force routing is not required here, caller will do it anyway
@@ -3512,6 +3512,22 @@
if (16 == bit_width) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
+
+ //check if mulitchannel clip needs to be down sampled to 48k
+ property_get("audio.playback.mch.downsample",value,"");
+ if (!strncmp("true", value, sizeof("true"))) {
+ out = usecase->stream.out;
+ if ((popcount(out->channel_mask) > 2) &&
+ (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+ !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ /* update out sample rate to reflect current backend sample rate */
+ out->sample_rate = sample_rate;
+ ALOGD("%s: MCH session defaulting sample rate to %d",
+ __func__, sample_rate);
+ }
+ }
+
// 24 bit playback on speakers is allowed through 48 khz backend only
// bit width re-configured based on platform info
if ((24 == bit_width) &&
@@ -3798,7 +3814,7 @@
strlcpy(mixer_ctl_name, "Playback Channel Map", sizeof(mixer_ctl_name));
if (snd_id >= 0) {
snprintf(device_num, sizeof(device_num), "%d", snd_id);
- strncat(mixer_ctl_name, device_num, 13);
+ strlcat(mixer_ctl_name, device_num, sizeof(device_num));
}
ALOGD("%s mixer_ctl_name:%s", __func__, mixer_ctl_name);
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 6d5b4a0..78c9ff8 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -220,17 +220,6 @@
#define SPKR_PROT_CALIB_TX_PCM_DEVICE 26
#define PLAYBACK_OFFLOAD_DEVICE 9
-#ifdef MULTIPLE_OFFLOAD_ENABLED
-#define PLAYBACK_OFFLOAD_DEVICE2 17
-#define PLAYBACK_OFFLOAD_DEVICE3 18
-#define PLAYBACK_OFFLOAD_DEVICE4 37
-#define PLAYBACK_OFFLOAD_DEVICE5 38
-#define PLAYBACK_OFFLOAD_DEVICE6 39
-#define PLAYBACK_OFFLOAD_DEVICE7 40
-#define PLAYBACK_OFFLOAD_DEVICE8 41
-#define PLAYBACK_OFFLOAD_DEVICE9 42
-#endif
-
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
/* Define macro for Internal FM volume mixer */
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index cc2d962..dbfc8ff 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -2647,6 +2647,11 @@
bool status = false;
str_parms_del(parms, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE);
event_name = strtok_r(value, ",", &status_str);
+ if (!event_name) {
+ ret = -EINVAL;
+ ALOGE("%s: event_name is NULL", __func__);
+ goto done;
+ }
ALOGV("%s: recieved update of external audio device %s %s",
__func__,
event_name, status_str);
@@ -3132,6 +3137,7 @@
struct stream_out *out = NULL;
unsigned int bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
unsigned int sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ char value[PROPERTY_VALUE_MAX] = {0};
// For voice calls use default configuration
// force routing is not required here, caller will do it anyway
@@ -3171,6 +3177,21 @@
if (16 == bit_width) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
+
+ //check if mulitchannel clip needs to be down sampled to 48k
+ property_get("audio.playback.mch.downsample",value,"");
+ if (!strncmp("true", value, sizeof("true"))) {
+ out = usecase->stream.out;
+ if ((popcount(out->channel_mask) > 2) &&
+ (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+ !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
+ /* update out sample rate to reflect current backend sample rate */
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s: MCH session defaulting sample rate to %d",
+ __func__, sample_rate);
+ }
+ }
+
// 24 bit playback on speakers is allowed through 48 khz backend only
// bit width re-configured based on platform info
if ((24 == bit_width) &&
@@ -3440,7 +3461,7 @@
int platform_set_channel_map(void *platform, int ch_count, char *ch_map, int snd_id)
{
struct mixer_ctl *ctl;
- char mixer_ctl_name[44]; // max length of name is 44 as defined
+ char mixer_ctl_name[44] = {0}; // max length of name is 44 as defined
int ret;
unsigned int i;
int set_values[8] = {0};
diff --git a/post_proc/effect_api.c b/post_proc/effect_api.c
index e15db17..2fb79ed 100644
--- a/post_proc/effect_api.c
+++ b/post_proc/effect_api.c
@@ -110,11 +110,11 @@
*mixer = mixer_open(card);
if (!(*mixer)) {
ALOGE("Failed to open mixer");
- ctl = NULL;
+ *ctl = NULL;
return -EINVAL;
} else {
*ctl = mixer_get_ctl_by_name(*mixer, mixer_string);
- if (!(*ctl)) {
+ if (!*ctl) {
ALOGE("mixer_get_ctl_by_name failed");
mixer_close(*mixer);
*mixer = NULL;