Merge "hal: Add support for VoWLAN feature"
diff --git a/hal/Android.mk b/hal/Android.mk
index 1c3f946..32a9ee9 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -121,7 +121,8 @@
libtinycompress \
libaudioroute \
libdl \
- libexpat
+ libexpat \
+ libmdmdetect
LOCAL_C_INCLUDES += \
external/tinyalsa/include \
@@ -131,7 +132,8 @@
$(call include-path-for, audio-effects) \
$(LOCAL_PATH)/$(AUDIO_PLATFORM) \
$(LOCAL_PATH)/audio_extn \
- $(LOCAL_PATH)/voice_extn
+ $(LOCAL_PATH)/voice_extn \
+ $(TARGET_OUT_HEADERS)/libmdmdetect/inc
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_LISTEN)),true)
LOCAL_CFLAGS += -DAUDIO_LISTEN_ENABLED
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index c6a8e68..245f5ca 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -174,7 +174,7 @@
#endif /* ANC_HEADSET_ENABLED */
#ifndef AFE_PROXY_ENABLED
-#define audio_extn_set_afe_proxy_parameters(parms) (0)
+#define audio_extn_set_afe_proxy_parameters(adev, parms) (0)
#define audio_extn_get_afe_proxy_parameters(query, reply) (0)
#else
/* Front left channel. */
@@ -290,7 +290,8 @@
return ret;
}
-void audio_extn_set_afe_proxy_parameters(struct str_parms *parms)
+void audio_extn_set_afe_proxy_parameters(struct audio_device *adev,
+ struct str_parms *parms)
{
int ret, val;
char value[32]={0};
@@ -300,6 +301,7 @@
if (ret >= 0) {
val = atoi(value);
aextnmod.proxy_channel_num = val;
+ adev->cur_wfd_channels = val;
ALOGD("%s: channel capability set to: %d", __func__,
aextnmod.proxy_channel_num);
}
@@ -358,7 +360,7 @@
struct str_parms *parms)
{
audio_extn_set_anc_parameters(adev, parms);
- audio_extn_set_afe_proxy_parameters(parms);
+ audio_extn_set_afe_proxy_parameters(adev, parms);
audio_extn_fm_set_parameters(adev, parms);
audio_extn_listen_set_parameters(adev, parms);
audio_extn_hfp_set_parameters(adev, parms);
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index fb428db..d432670 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -184,8 +184,10 @@
#ifndef HFP_ENABLED
#define audio_extn_hfp_is_active(adev) (0)
+#define audio_extn_hfp_get_usecase() (0)
#else
bool audio_extn_hfp_is_active(struct audio_device *adev);
+audio_usecase_t audio_extn_hfp_get_usecase();
#endif
#endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/dolby.c b/hal/audio_extn/dolby.c
index bcc7381..99fa2b7 100644
--- a/hal/audio_extn/dolby.c
+++ b/hal/audio_extn/dolby.c
@@ -64,7 +64,7 @@
/* DS1-DDP Endp Params */
#define DDP_ENDP_NUM_PARAMS 17
-#define DDP_ENDP_NUM_DEVICES 22
+#define DDP_ENDP_NUM_DEVICES 23
static int ddp_endp_params_id[DDP_ENDP_NUM_PARAMS] = {
PARAM_ID_MAX_OUTPUT_CHANNELS, PARAM_ID_CTL_RUNNING_MODE,
PARAM_ID_CTL_ERROR_CONCEAL, PARAM_ID_CTL_ERROR_MAX_RPTS,
@@ -147,7 +147,10 @@
{8, 0, 0, 0, 0, 0, 0, 21, 1, 6, 0, 0, 0, 0, 0, 0, 0},
{1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
{AUDIO_DEVICE_OUT_PROXY, 2,
- {8, 0, 0, 0, 0, 0, 0, 21, 1, 6, 0, 0, 0, 0, 0, 0, 0},
+ {8, 0, 0, 0, 0, 0, 0, 21, 1, 2, 0, 0, 0, 0, 0, 0, 0},
+ {1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
+ {AUDIO_DEVICE_OUT_PROXY, 6,
+ {8, 0, 0, 0, 0, 0, 0, 21, 1, 2, 0, 0, 0, 0, 0, 0, 0},
{1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
};
@@ -264,9 +267,16 @@
(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
((usecase->stream.out->format == AUDIO_FORMAT_AC3) ||
(usecase->stream.out->format == AUDIO_FORMAT_EAC3))) {
+ /*
+ * Use wfd /hdmi sink channel cap for dolby params if device is wfd
+ * or hdmi. Otherwise use stereo configuration
+ */
+ int channel_cap = usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
+ adev->cur_hdmi_channels :
+ usecase->devices & AUDIO_DEVICE_OUT_PROXY ?
+ adev->cur_wfd_channels : 2;
send_ddp_endp_params_stream(usecase->stream.out, usecase->devices,
- usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
- adev->cur_hdmi_channels : 2, false /* set cache */);
+ channel_cap, false /* set cache */);
}
}
}
@@ -334,7 +344,9 @@
update_ddp_endp_table(ddp_dev, dev_ch_cap,
PARAM_ID_OUT_CTL_STEREO_MODE, val);
}
-
+ /* TODO: Do we need device channel caps here?
+ * We dont have that information as this is from dolby modules
+ */
send_ddp_endp_params(adev, ddp_dev, dev_ch_cap);
}
@@ -343,13 +355,20 @@
audio_format_t format)
{
int id = 0;
+ /*
+ * Use wfd /hdmi sink channel cap for dolby params if device is wfd
+ * or hdmi. Otherwise use stereo configuration
+ */
+ int channel_cap = out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
+ adev->cur_hdmi_channels :
+ out->devices & AUDIO_DEVICE_OUT_PROXY ?
+ adev->cur_wfd_channels : 2;
switch (format) {
case AUDIO_FORMAT_AC3:
id = SND_AUDIOCODEC_AC3;
send_ddp_endp_params_stream(out, out->devices,
- out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
- adev->cur_hdmi_channels : 2, true /* set_cache */);
+ channel_cap, true /* set_cache */);
#ifndef DS1_DOLBY_DAP_ENABLED
audio_extn_dolby_set_dmid(adev);
#endif
@@ -357,8 +376,7 @@
case AUDIO_FORMAT_EAC3:
id = SND_AUDIOCODEC_EAC3;
send_ddp_endp_params_stream(out, out->devices,
- out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
- adev->cur_hdmi_channels : 2, true /* set_cache */);
+ channel_cap, true /* set_cache */);
#ifndef DS1_DOLBY_DAP_ENABLED
audio_extn_dolby_set_dmid(adev);
#endif
@@ -430,8 +448,7 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if ((usecase->type == PCM_PLAYBACK) &&
- (usecase->id != USECASE_AUDIO_PLAYBACK_LOW_LATENCY))
+ if (usecase->type == PCM_PLAYBACK)
send = true;
}
if (!send)
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 2d6e1e0..c480490 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -220,7 +220,7 @@
bool audio_extn_hfp_is_active(struct audio_device *adev)
{
struct audio_usecase *hfp_usecase = NULL;
- hfp_usecase = get_usecase_from_list(adev, USECASE_AUDIO_HFP_SCO);
+ hfp_usecase = get_usecase_from_list(adev, hfpmod.ucid);
if (hfp_usecase != NULL)
return true;
@@ -228,6 +228,11 @@
return false;
}
+audio_usecase_t audio_extn_hfp_get_usecase()
+{
+ return hfpmod.ucid;
+}
+
void audio_extn_hfp_set_parameters(struct audio_device *adev, struct str_parms *parms)
{
int ret;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 5514007..69e9561 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -603,6 +603,7 @@
struct audio_usecase *vc_usecase = NULL;
struct audio_usecase *voip_usecase = NULL;
struct audio_usecase *hfp_usecase = NULL;
+ audio_usecase_t hfp_ucid;
struct listnode *node;
int status = 0;
@@ -642,7 +643,8 @@
out_snd_device = voip_usecase->out_snd_device;
}
} else if (audio_extn_hfp_is_active(adev)) {
- hfp_usecase = get_usecase_from_list(adev, USECASE_AUDIO_HFP_SCO);
+ hfp_ucid = audio_extn_hfp_get_usecase();
+ hfp_usecase = get_usecase_from_list(adev, hfp_ucid);
if (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
in_snd_device = hfp_usecase->in_snd_device;
out_snd_device = hfp_usecase->out_snd_device;
@@ -921,24 +923,30 @@
send_callback = false;
switch(cmd->cmd) {
case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+ ALOGD("copl(%x):calling compress_wait", (unsigned int)out);
compress_wait(out->compr, -1);
+ ALOGD("copl(%x):out of compress_wait", (unsigned int)out);
send_callback = true;
event = STREAM_CBK_EVENT_WRITE_READY;
break;
case OFFLOAD_CMD_PARTIAL_DRAIN:
ret = compress_next_track(out->compr);
- if(ret == 0)
+ if(ret == 0) {
+ ALOGD("copl(%x):calling compress_partial_drain", (unsigned int)out);
compress_partial_drain(out->compr);
+ ALOGD("copl(%x):out of compress_partial_drain", (unsigned int)out);
+ }
else if(ret == -ETIMEDOUT)
compress_drain(out->compr);
else
ALOGE("%s: Next track returned error %d",__func__, ret);
-
send_callback = true;
event = STREAM_CBK_EVENT_DRAIN_READY;
break;
case OFFLOAD_CMD_DRAIN:
+ ALOGD("copl(%x):calling compress_drain", (unsigned int)out);
compress_drain(out->compr);
+ ALOGD("copl(%x):calling compress_drain", (unsigned int)out);
send_callback = true;
event = STREAM_CBK_EVENT_DRAIN_READY;
break;
@@ -1013,6 +1021,12 @@
"no change in HDMI channels", __func__);
ret = false;
break;
+ } else if (usecase->id == USECASE_AUDIO_PLAYBACK_OFFLOAD &&
+ popcount(usecase->stream.out->channel_mask) > 2) {
+ ALOGD("%s: multi-channel(%x) compress offload playback is active, "
+ "no change in HDMI channels", __func__, usecase->stream.out->channel_mask);
+ ret = false;
+ break;
}
}
}
@@ -1110,7 +1124,7 @@
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
- ALOGV("%s: enter: usecase(%d: %s) devices(%#x)",
+ ALOGD("%s: enter: usecase(%d: %s) devices(%#x)",
__func__, out->usecase, use_case_table[out->usecase], out->devices);
out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
if (out->pcm_device_id < 0) {
@@ -1312,6 +1326,7 @@
out->pcm = NULL;
}
} else {
+ ALOGD("copl(%x):standby", (unsigned int)out);
stop_compressed_output_l(out);
out->gapless_mdata.encoder_delay = 0;
out->gapless_mdata.encoder_padding = 0;
@@ -1595,9 +1610,9 @@
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
- ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes);
+ ALOGD("copl(%x): writing buffer (%d bytes) to compress device", (unsigned int)out, bytes);
if (out->send_new_metadata) {
- ALOGVV("send new gapless metadata");
+ ALOGD("copl(%x):send new gapless metadata", (unsigned int)out);
compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
out->send_new_metadata = 0;
}
@@ -1605,6 +1620,7 @@
ret = compress_write(out->compr, buffer, bytes);
ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __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);
}
if (!out->playback_started) {
@@ -1737,6 +1753,7 @@
int status = -ENOSYS;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+ ALOGD("copl(%x):pause compress driver", (unsigned int)out);
pthread_mutex_lock(&out->lock);
if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
status = compress_pause(out->compr);
@@ -1753,6 +1770,7 @@
int status = -ENOSYS;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+ ALOGD("copl(%x):resume compress driver", (unsigned int)out);
status = 0;
pthread_mutex_lock(&out->lock);
if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
@@ -1785,9 +1803,11 @@
struct stream_out *out = (struct stream_out *)stream;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+ ALOGD("copl(%x):calling compress flush", (unsigned int)out);
pthread_mutex_lock(&out->lock);
stop_compressed_output_l(out);
pthread_mutex_unlock(&out->lock);
+ ALOGD("copl(%x):out of compress flush", (unsigned int)out);
return 0;
}
return -ENOSYS;
@@ -2126,6 +2146,9 @@
goto error_open;
}
} else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ ALOGD("%s: copl(%x): sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
+ __func__, (unsigned int)out, config->sample_rate, config->channel_mask, devices, flags);
+
if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
ALOGE("%s: Unsupported Offload information", __func__);
@@ -2673,6 +2696,7 @@
adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
voice_init(adev);
list_init(&adev->usecase_list);
+ adev->cur_wfd_channels = 2;
/* Loads platform specific libraries dynamically */
adev->platform = platform_init(adev);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index a6824c1..2108a00 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -224,6 +224,7 @@
bool speaker_lr_swap;
struct voice voice;
unsigned int cur_hdmi_channels;
+ unsigned int cur_wfd_channels;
int snd_card;
void *platform;
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 128e4af..59bdb56 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -125,7 +125,9 @@
static void update_hardware_info_8084(struct hardware_info *hw_info, const char *snd_card_name)
{
- if (!strcmp(snd_card_name, "apq8084-taiko-mtp-snd-card")) {
+ if (!strcmp(snd_card_name, "apq8084-taiko-mtp-snd-card") ||
+ !strncmp(snd_card_name, "apq8084-taiko-i2s-mtp-snd-card",
+ sizeof("apq8084-taiko-i2s-mtp-snd-card"))) {
strlcpy(hw_info->type, "mtp", sizeof(hw_info->type));
strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
hw_info->snd_devices = NULL;
@@ -137,6 +139,13 @@
hw_info->snd_devices = (snd_device_t *)taiko_apq8084_CDP_variant_devices;
hw_info->num_snd_devices = ARRAY_SIZE(taiko_apq8084_CDP_variant_devices);
strlcpy(hw_info->dev_extn, "-cdp", sizeof(hw_info->dev_extn));
+ } else if (!strncmp(snd_card_name, "apq8084-taiko-i2s-cdp-snd-card",
+ sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
+ strlcpy(hw_info->type, " cdp", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
+ hw_info->snd_devices = NULL;
+ hw_info->num_snd_devices = 0;
+ strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
} else if (!strcmp(snd_card_name, "apq8084-taiko-liquid-snd-card")) {
strlcpy(hw_info->type , " liquid", sizeof(hw_info->type));
strlcpy(hw_info->name, "apq8084", sizeof(hw_info->type));
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index c727900..684436e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -32,9 +32,15 @@
#include "audio_extn.h"
#include "voice_extn.h"
#include "sound/compress_params.h"
+#include "mdm_detect.h"
#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
#define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
+#define MIXER_XML_PATH_I2S "/system/etc/mixer_paths_i2s.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 LIB_ACDB_LOADER "libacdbloader.so"
#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
@@ -108,6 +114,7 @@
int fluence_type;
int btsco_sample_rate;
bool slowtalk;
+ bool is_i2s_ext_modem;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -410,7 +417,7 @@
return 0;
}
-static struct csd_data *open_csd_client()
+static struct csd_data *open_csd_client(bool i2s_ext_modem)
{
struct csd_data *csd = calloc(1, sizeof(struct csd_data));
@@ -512,6 +519,16 @@
__func__, dlerror());
goto error;
}
+
+ csd->get_sample_rate = (get_sample_rate_t)dlsym(csd->csd_client,
+ "csd_client_get_sample_rate");
+ if (csd->get_sample_rate == NULL) {
+ ALOGE("%s: dlsym error %s for csd_client_get_sample_rate",
+ __func__, dlerror());
+
+ goto error;
+ }
+
csd->init = (init_t)dlsym(csd->csd_client, "csd_client_init");
if (csd->init == NULL) {
@@ -519,7 +536,7 @@
__func__, dlerror());
goto error;
} else {
- csd->init();
+ csd->init(i2s_ext_modem);
}
}
return csd;
@@ -540,10 +557,44 @@
}
}
+static void platform_csd_init(struct platform_data *plat_data)
+{
+ struct dev_info mdm_detect_info;
+ int ret = 0;
+
+ /* Call ESOC API to get the number of modems.
+ * If the number of modems is not zero, load CSD Client specific
+ * symbols. Voice call is handled by MDM and apps processor talks to
+ * MDM through CSD Client
+ */
+ ret = get_system_info(&mdm_detect_info);
+ if (ret > 0) {
+ ALOGE("%s: Failed to get system info, ret %d", __func__, ret);
+ }
+ ALOGD("%s: num_modems %d\n", __func__, mdm_detect_info.num_modems);
+
+ if (mdm_detect_info.num_modems > 0)
+ plat_data->csd = open_csd_client(plat_data->is_i2s_ext_modem);
+}
+
+static bool platform_is_i2s_ext_modem(const char *snd_card_name,
+ struct platform_data *plat_data)
+{
+ plat_data->is_i2s_ext_modem = false;
+
+ if (!strncmp(snd_card_name, "apq8084-taiko-i2s-mtp-snd-card",
+ sizeof("apq8084-taiko-i2s-mtp-snd-card")) ||
+ !strncmp(snd_card_name, "apq8084-taiko-i2s-cdp-snd-card",
+ sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
+ plat_data->is_i2s_ext_modem = true;
+ }
+ ALOGV("%s, is_i2s_ext_modem:%d",__func__, plat_data->is_i2s_ext_modem);
+
+ return plat_data->is_i2s_ext_modem;
+}
+
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;
@@ -575,10 +626,16 @@
if (!my_data->hw_info) {
ALOGE("%s: Failed to init hardware info", __func__);
} else {
- if (audio_extn_read_xml(adev, snd_card_num, MIXER_XML_PATH,
- MIXER_XML_PATH_AUXPCM) == -ENOSYS)
+ if (platform_is_i2s_ext_modem(snd_card_name, my_data)) {
+ ALOGD("%s: Call MIXER_XML_PATH_I2S", __func__);
+
+ adev->audio_route = audio_route_init(snd_card_num,
+ MIXER_XML_PATH_I2S);
+ } else if (audio_extn_read_xml(adev, snd_card_num, MIXER_XML_PATH,
+ MIXER_XML_PATH_AUXPCM) == -ENOSYS) {
adev->audio_route = audio_route_init(snd_card_num,
MIXER_XML_PATH);
+ }
if (!adev->audio_route) {
ALOGE("%s: Failed to init audio route controls, aborting.",
__func__);
@@ -677,18 +734,13 @@
}
/* Initialize ACDB ID's */
- platform_info_init();
+ if (my_data->is_i2s_ext_modem)
+ platform_info_init(PLATFORM_INFO_XML_PATH_I2S);
+ else
+ platform_info_init(PLATFORM_INFO_XML_PATH);
- /* If platform is apq8084 and baseband is MDM, load CSD Client specific
- * symbols. Voice call is handled by MDM and apps processor talks to
- * MDM through CSD Client
- */
- property_get("ro.board.platform", platform, "");
- property_get("ro.baseband", baseband, "");
- if (!strncmp("apq8084", platform, sizeof("apq8084")) &&
- !strncmp("mdm", baseband, sizeof("mdm"))) {
- my_data->csd = open_csd_client();
- }
+ /* load csd client */
+ platform_csd_init(my_data);
/* init usb */
audio_extn_usb_init(adev);
@@ -961,6 +1013,20 @@
return ret;
}
+int platform_get_sample_rate(void *platform, uint32_t *rate)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ int ret = 0;
+
+ if ((my_data->csd != NULL) && my_data->is_i2s_ext_modem) {
+ ret = my_data->csd->get_sample_rate(rate);
+ if (ret < 0) {
+ ALOGE("%s: csd_get_sample_rate error %d\n", __func__, ret);
+ }
+ }
+ return ret;
+}
+
int platform_set_voice_volume(void *platform, int volume)
{
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index d9b4302..0f72a54 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -225,7 +225,7 @@
#define LIB_CSD_CLIENT "libcsd-client.so"
/* CSD-CLIENT related functions */
-typedef int (*init_t)();
+typedef int (*init_t)(bool);
typedef int (*deinit_t)();
typedef int (*disable_device_t)();
typedef int (*enable_device_config_t)(int, int);
@@ -239,6 +239,7 @@
typedef int (*stop_playback_t)(uint32_t);
typedef int (*start_record_t)(uint32_t, int);
typedef int (*stop_record_t)(uint32_t);
+typedef int (*get_sample_rate_t)(uint32_t *);
/* CSD Client structure */
struct csd_data {
void *csd_client;
@@ -256,6 +257,7 @@
stop_playback_t stop_playback;
start_record_t start_record;
stop_record_t stop_record;
+ get_sample_rate_t get_sample_rate;
};
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 81291a2..2ec6b50 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -44,6 +44,7 @@
int platform_stop_voice_call(void *platform, uint32_t vsid);
int platform_set_voice_volume(void *platform, int volume);
int platform_set_mic_mute(void *platform, bool state);
+int platform_get_sample_rate(void *platform, uint32_t *rate);
snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices);
snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device);
int platform_set_hdmi_channels(void *platform, int channel_count);
@@ -63,7 +64,7 @@
bool platform_listen_update_status(snd_device_t snd_device);
/* From platform_info_parser.c */
-int platform_info_init(void);
+int platform_info_init(const char *filename);
struct audio_offload_info_t;
uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 8f56107..85a05eb 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -38,7 +38,6 @@
#include "platform_api.h"
#include <platform.h>
-#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define BUF_SIZE 1024
static void process_device(const XML_Char **attr)
@@ -52,20 +51,20 @@
index = platform_get_snd_device_index((char *)attr[1]);
if (index < 0) {
- ALOGE("%s: Device %s in %s not found, no ACDB ID set!",
- __func__, attr[1], PLATFORM_INFO_XML_PATH);
+ ALOGE("%s: Device %s in platform info xml not found, no ACDB ID set!",
+ __func__, attr[1]);
goto done;
}
if (strcmp(attr[2], "acdb_id") != 0) {
- ALOGE("%s: Device %s in %s has no acdb_id, no ACDB ID set!",
- __func__, attr[1], PLATFORM_INFO_XML_PATH);
+ ALOGE("%s: Device %s in platform info xml has no acdb_id, no ACDB ID set!",
+ __func__, attr[1]);
goto done;
}
if(platform_set_snd_device_acdb_id(index, atoi((char *)attr[3])) < 0) {
- ALOGE("%s: Device %s in %s, ACDB ID %d was not set!",
- __func__, attr[1], PLATFORM_INFO_XML_PATH, atoi((char *)attr[3]));
+ ALOGE("%s: Device %s in platform info xml ACDB ID %d was not set!",
+ __func__, attr[1], atoi((char *)attr[3]));
goto done;
}
@@ -91,7 +90,7 @@
}
-int platform_info_init(void)
+int platform_info_init(const char *filename)
{
XML_Parser parser;
FILE *file;
@@ -99,10 +98,10 @@
int bytes_read;
void *buf;
- file = fopen(PLATFORM_INFO_XML_PATH, "r");
+ file = fopen(filename, "r");
if (!file) {
ALOGD("%s: Failed to open %s, using defaults.",
- __func__, PLATFORM_INFO_XML_PATH);
+ __func__, filename);
ret = -ENODEV;
goto done;
}
@@ -134,7 +133,7 @@
if (XML_ParseBuffer(parser, bytes_read,
bytes_read == 0) == XML_STATUS_ERROR) {
ALOGE("%s: XML_ParseBuffer failed, for %s",
- __func__, PLATFORM_INFO_XML_PATH);
+ __func__, filename);
ret = -EINVAL;
goto err_free_parser;
}
diff --git a/hal/voice.c b/hal/voice.c
index 28d44db..1c3ab38 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -106,6 +106,7 @@
int i, ret = 0;
struct audio_usecase *uc_info;
int pcm_dev_rx_id, pcm_dev_tx_id;
+ uint32_t sample_rate = 8000;
struct voice_session *session = NULL;
struct pcm_config voice_config = pcm_config_voice_call;
@@ -133,6 +134,13 @@
ret = -EIO;
goto error_start_voice;
}
+ ret = platform_get_sample_rate(adev->platform, &sample_rate);
+ if (ret < 0) {
+ ALOGE("platform_get_sample_rate error %d\n", ret);
+ } else {
+ voice_config.rate = sample_rate;
+ }
+ ALOGD("voice_config.rate %d\n", voice_config.rate);
ALOGV("%s: Opening PCM playback device card_id(%d) device_id(%d)",
__func__, adev->snd_card, pcm_dev_rx_id);
diff --git a/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h b/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
index 276eaa3..623caa8 100644
--- a/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
+++ b/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
@@ -1,5 +1,5 @@
/*--------------------------------------------------------------------------
-Copyright (c) 2010, The Linux Foundation. All rights reserved.
+Copyright (c) 2010-2014, 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 met:
@@ -358,7 +358,10 @@
OMX_COMPONENT_OUTPUT_DISABLE_PENDING =0x7
};
-
+ #define MIN_BITRATE 24000
+ #define MAX_BITRATE 192000
+ #define MAX_BITRATE_MULFACTOR 12
+ #define BITRATE_DIVFACTOR 2
typedef Map<OMX_BUFFERHEADERTYPE*, OMX_BUFFERHEADERTYPE*>
input_buffer_map;
@@ -619,6 +622,7 @@
OMX_U8 num_bits_reqd,
OMX_U32 value,
OMX_U16 *hdr_bit_index);
+ int get_updated_bit_rate(int bitrate);
};
#endif
diff --git a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
index 52aa915..6521265 100644
--- a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
+++ b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
@@ -1,5 +1,5 @@
/*--------------------------------------------------------------------------
-Copyright (c) 2010, The Linux Foundation. All rights reserved.
+Copyright (c) 2010-2014, 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 met:
@@ -1438,10 +1438,12 @@
}
drv_aac_enc_config.channels = m_aac_param.nChannels;
drv_aac_enc_config.sample_rate = m_aac_param.nSampleRate;
- drv_aac_enc_config.bit_rate = m_aac_param.nBitRate;
- DEBUG_PRINT("aac config %lu,%lu,%lu %d\n",
+ drv_aac_enc_config.bit_rate =
+ get_updated_bit_rate(m_aac_param.nBitRate);
+ DEBUG_PRINT("aac config %lu,%lu,%lu %d updated bitrate %d\n",
m_aac_param.nChannels,m_aac_param.nSampleRate,
- m_aac_param.nBitRate,m_aac_param.eAACStreamFormat);
+ m_aac_param.nBitRate,m_aac_param.eAACStreamFormat,
+ drv_aac_enc_config.bit_rate);
switch(m_aac_param.eAACStreamFormat)
{
@@ -5014,3 +5016,44 @@
}
+int omx_aac_aenc::get_updated_bit_rate(int bitrate)
+{
+ int updated_rate, min_bitrate, max_bitrate;
+
+ max_bitrate = m_aac_param.nSampleRate *
+ MAX_BITRATE_MULFACTOR;
+ switch(m_aac_param.eAACProfile)
+ {
+ case OMX_AUDIO_AACObjectLC:
+ min_bitrate = m_aac_param.nSampleRate;
+ if (m_aac_param.nChannels == 1) {
+ min_bitrate = min_bitrate/BITRATE_DIVFACTOR;
+ max_bitrate = max_bitrate/BITRATE_DIVFACTOR;
+ }
+ break;
+ case OMX_AUDIO_AACObjectHE:
+ min_bitrate = MIN_BITRATE;
+ if (m_aac_param.nChannels == 1)
+ max_bitrate = max_bitrate/BITRATE_DIVFACTOR;
+ break;
+ case OMX_AUDIO_AACObjectHE_PS:
+ min_bitrate = MIN_BITRATE;
+ break;
+ default:
+ return bitrate;
+ break;
+ }
+ /* Update MIN and MAX values*/
+ if (min_bitrate > MIN_BITRATE)
+ min_bitrate = MIN_BITRATE;
+ if (max_bitrate > MAX_BITRATE)
+ max_bitrate = MAX_BITRATE;
+ /* Update the bitrate in the range */
+ if (bitrate < min_bitrate)
+ updated_rate = min_bitrate;
+ else if(bitrate > max_bitrate)
+ updated_rate = max_bitrate;
+ else
+ updated_rate = bitrate;
+ return updated_rate;
+}
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index a925e8e..9e6c1fc 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -211,9 +211,14 @@
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))) {
offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), true);
+ if (bass_ctxt->ctl && bass_ctxt->strength)
+ offload_bassboost_send_params(bass_ctxt->ctl,
+ bass_ctxt->offload_bass,
+ OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG |
+ OFFLOAD_SEND_BASSBOOST_STRENGTH);
+ }
return 0;
}
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 9682b93..4190129 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -210,9 +210,14 @@
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))) {
offload_virtualizer_set_enable_flag(&(virt_ctxt->offload_virt), true);
+ if (virt_ctxt->ctl && virt_ctxt->strength)
+ offload_virtualizer_send_params(virt_ctxt->ctl,
+ virt_ctxt->offload_virt,
+ OFFLOAD_SEND_VIRTUALIZER_ENABLE_FLAG |
+ OFFLOAD_SEND_BASSBOOST_STRENGTH);
+ }
return 0;
}