Merge "mm-audio: Update bitrate based on profile and samplerate"
diff --git a/Android.mk b/Android.mk
index 9bb3250..c63df05 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916,$(TARGET_BOARD_PLATFORM)),)
MY_LOCAL_PATH := $(call my-dir)
diff --git a/hal/Android.mk b/hal/Android.mk
index a7e0a02..32a9ee9 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -8,7 +8,7 @@
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter msm8974 msm8226 msm8610 apq8084,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8610 apq8084 msm8916,$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM = msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -23,6 +23,7 @@
endif
endif
+
LOCAL_SRC_FILES := \
audio_hw.c \
voice.c \
@@ -53,6 +54,10 @@
LOCAL_SRC_FILES += audio_extn/hfp.c
endif
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_CUSTOMSTEREO)),true)
+ LOCAL_CFLAGS += -DCUSTOM_STEREO_ENABLED
+endif
+
ifneq ($(strip $(AUDIO_FEATURE_DISABLED_SSR)),true)
LOCAL_CFLAGS += -DSSR_ENABLED
LOCAL_SRC_FILES += audio_extn/ssr.c
@@ -116,7 +121,8 @@
libtinycompress \
libaudioroute \
libdl \
- libexpat
+ libexpat \
+ libmdmdetect
LOCAL_C_INCLUDES += \
external/tinyalsa/include \
@@ -126,7 +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 ad487b1..245f5ca 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -35,18 +35,22 @@
struct audio_extn_module {
bool anc_enabled;
bool aanc_enabled;
+ bool custom_stereo_enabled;
uint32_t proxy_channel_num;
};
static struct audio_extn_module aextnmod = {
.anc_enabled = 0,
.aanc_enabled = 0,
+ .custom_stereo_enabled = 0,
.proxy_channel_num = 2,
};
#define AUDIO_PARAMETER_KEY_ANC "anc_enabled"
#define AUDIO_PARAMETER_KEY_WFD "wfd_channel_cap"
#define AUDIO_PARAMETER_CAN_OPEN_PROXY "can_open_proxy"
+#define AUDIO_PARAMETER_CUSTOM_STEREO "stereo_as_dual_mono"
+
#ifndef FM_ENABLED
#define audio_extn_fm_set_parameters(adev, parms) (0)
#else
@@ -60,6 +64,45 @@
struct str_parms *parms);
#endif
+#ifndef CUSTOM_STEREO_ENABLED
+#define audio_extn_customstereo_set_parameters(adev, parms) (0)
+#else
+void audio_extn_customstereo_set_parameters(struct audio_device *adev,
+ struct str_parms *parms)
+{
+ int ret = 0;
+ char value[32]={0};
+ bool custom_stereo_state = false;
+ const char *mixer_ctl_name = "Set Custom Stereo OnOff";
+ struct mixer_ctl *ctl;
+
+ ALOGV("%s", __func__);
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_CUSTOM_STEREO, value,
+ sizeof(value));
+ if (ret >= 0) {
+ if (!strncmp("true", value, sizeof("true")) || atoi(value))
+ custom_stereo_state = true;
+
+ if (custom_stereo_state == aextnmod.custom_stereo_enabled)
+ return;
+
+ 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;
+ }
+ if (mixer_ctl_set_value(ctl, 0, custom_stereo_state) < 0) {
+ ALOGE("%s: Could not set custom stereo state %d",
+ __func__, custom_stereo_state);
+ return;
+ }
+ aextnmod.custom_stereo_enabled = custom_stereo_state;
+ ALOGV("%s: Setting custom stereo state success", __func__);
+ }
+}
+#endif /* CUSTOM_STEREO_ENABLED */
+
#ifndef ANC_HEADSET_ENABLED
#define audio_extn_set_anc_parameters(adev, parms) (0)
#else
@@ -131,7 +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. */
@@ -247,7 +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};
@@ -257,6 +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);
}
@@ -315,11 +360,12 @@
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);
audio_extn_ddp_set_parameters(adev, parms);
+ audio_extn_customstereo_set_parameters(adev, parms);
}
void audio_extn_get_parameters(const struct audio_device *adev,
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_extn/listen.c b/hal/audio_extn/listen.c
index 4a1980b..91bb04f 100644
--- a/hal/audio_extn/listen.c
+++ b/hal/audio_extn/listen.c
@@ -1,4 +1,4 @@
-/* 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
@@ -64,6 +64,7 @@
typedef void (*destroy_listen_hw_t)();
typedef int (*open_listen_session_t)(struct audio_hw_device *,
+ struct listen_open_params*,
struct listen_session**);
typedef int (*close_listen_session_t)(struct audio_hw_device *dev,
@@ -119,7 +120,6 @@
void audio_extn_listen_set_parameters(struct audio_device *adev,
struct str_parms *parms)
{
-
ALOGV("%s: enter", __func__);
if (listen_dev) {
char *kv_pairs = str_parms_to_str(parms);
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 88e3cad..699c3b7 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -152,7 +152,7 @@
file_size = st.st_size;
- read_buf = (char *)calloc(1, USB_BUFF_SIZE);
+ read_buf = (char *)calloc(1, USB_BUFF_SIZE + 1);
err = read(fd, read_buf, USB_BUFF_SIZE);
str_start = strstr(read_buf, type);
if (str_start == NULL) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 04a7aba..3bb9098 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -53,16 +53,14 @@
#include "voice_extn.h"
#include "sound/compress_params.h"
+#include "sound/asound.h"
-#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
-#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
-#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
-#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
struct pcm_config pcm_config_deep_buffer = {
@@ -155,35 +153,6 @@
static int set_voice_volume_l(struct audio_device *adev, float volume);
-/* Read offload buffer size from a property.
- * If value is not power of 2 round it to
- * power of 2.
- */
-static uint32_t get_offload_buffer_size(audio_offload_info_t* info)
-{
- char value[PROPERTY_VALUE_MAX] = {0};
- uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
- if((property_get("audio.offload.buffer.size.kb", value, "")) &&
- atoi(value)) {
- fragment_size = atoi(value) * 1024;
- //ring buffer size needs to be 4k aligned.
- CHECK(!(fragment_size * COMPRESS_OFFLOAD_NUM_FRAGMENTS % 4096));
- }
-
- if (info != NULL && info->has_video && info->is_streaming) {
- fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
- ALOGV("%s: offload fragment size reduced for AV streaming to %d",
- __func__, out->compr_config.fragment_size);
- }
-
- if(fragment_size < MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
- fragment_size = MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
- else if(fragment_size > MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
- fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
- ALOGVV("%s: fragment_size %d", __func__, fragment_size);
- return fragment_size;
-}
-
static int check_and_set_gapless_mode(struct audio_device *adev) {
@@ -214,8 +183,10 @@
static bool is_supported_format(audio_format_t format)
{
if (format == AUDIO_FORMAT_MP3 ||
- format == AUDIO_FORMAT_AAC)
- return true;
+ format == AUDIO_FORMAT_AAC ||
+ format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
+ format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
+ return true;
return false;
}
@@ -231,6 +202,10 @@
case AUDIO_FORMAT_AAC:
id = SND_AUDIOCODEC_AAC;
break;
+ case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
+ case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+ id = SND_AUDIOCODEC_PCM;
+ break;
default:
ALOGE("%s: Unsupported audio format :%x", __func__, format);
}
@@ -627,6 +602,7 @@
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;
@@ -666,7 +642,8 @@
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;
@@ -729,6 +706,15 @@
disable_snd_device(adev, usecase->in_snd_device, false);
}
+ /* Applicable only on the targets that has external modem.
+ * New device information should be sent to modem before enabling
+ * the devices to reduce in-call device switch time.
+ */
+ if (usecase->type == VOICE_CALL)
+ status = platform_switch_voice_call_enable_device_config(adev->platform,
+ out_snd_device,
+ in_snd_device);
+
/* Enable new sound devices */
if (out_snd_device != SND_DEVICE_NONE) {
if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
@@ -936,24 +922,30 @@
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;
@@ -1028,6 +1020,12 @@
"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;
}
}
}
@@ -1125,7 +1123,7 @@
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) {
@@ -1327,6 +1325,7 @@
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;
@@ -1610,9 +1609,9 @@
}
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;
}
@@ -1620,6 +1619,7 @@
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) {
@@ -1752,6 +1752,7 @@
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);
@@ -1768,6 +1769,7 @@
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) {
@@ -1800,9 +1802,11 @@
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;
@@ -1914,11 +1918,10 @@
(voice_extn_compress_voip_is_format_supported(in->format)) &&
(in->config.rate == 8000 || in->config.rate == 16000) &&
(popcount(in->channel_mask) == 1)) {
- ret = voice_extn_compress_voip_open_input_stream(in);
- if (ret != 0) {
+ err = voice_extn_compress_voip_open_input_stream(in);
+ if (err != 0) {
ALOGE("%s: Compress voip input cannot be opened, error:%d",
- __func__, ret);
- goto done;
+ __func__, err);
}
}
}
@@ -2142,6 +2145,9 @@
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__);
@@ -2161,8 +2167,10 @@
out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
if (config->offload_info.channel_mask)
out->channel_mask = config->offload_info.channel_mask;
- else if (config->channel_mask)
+ else if (config->channel_mask) {
out->channel_mask = config->channel_mask;
+ config->offload_info.channel_mask = config->channel_mask;
+ }
out->format = config->offload_info.format;
out->sample_rate = config->offload_info.sample_rate;
@@ -2179,7 +2187,13 @@
else
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
- out->compr_config.fragment_size = get_offload_buffer_size(&config->offload_info);
+ if (audio_is_offload_pcm(config->offload_info.format)) {
+ out->compr_config.fragment_size =
+ platform_get_pcm_offload_buffer_size(&config->offload_info);
+ } else {
+ out->compr_config.fragment_size =
+ platform_get_compress_offload_buffer_size(&config->offload_info);
+ }
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
out->compr_config.codec->sample_rate =
compress_get_alsa_rate(config->offload_info.sample_rate);
@@ -2190,6 +2204,11 @@
out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
+ else if(config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
+
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
@@ -2676,6 +2695,7 @@
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 205977b..21b7790 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -223,6 +223,7 @@
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/platform.c b/hal/msm8974/platform.c
index c9719bb..4161d87 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -31,12 +31,31 @@
#include "platform.h"
#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 LIB_ACDB_LOADER "libacdbloader.so"
#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
+#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
+#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
+#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
+
+/* Used in calculating fragment size for pcm offload */
+#define PCM_OFFLOAD_BUFFER_DURATION_FOR_AV 2000 /* 2 secs */
+#define PCM_OFFLOAD_BUFFER_DURATION_FOR_AV_STREAMING 100 /* 100 millisecs */
+
+/* MAX PCM fragment size cannot be increased further due
+ * to flinger's cblk size of 1mb,and it has to be a multiple of
+ * 24 - lcm of channels supported by DSP
+ */
+#define MAX_PCM_OFFLOAD_FRAGMENT_SIZE (240 * 1024)
+#define MIN_PCM_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
+
+#define ALIGN( num, to ) (((num) + (to-1)) & (~(to-1)))
/*
* This file will have a maximum of 38 bytes:
*
@@ -416,6 +435,13 @@
__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) {
@@ -514,10 +540,28 @@
}
}
+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();
+}
+
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;
@@ -653,16 +697,8 @@
/* Initialize ACDB ID's */
platform_info_init();
- /* 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);
@@ -831,6 +867,32 @@
return ret;
}
+int platform_switch_voice_call_enable_device_config(void *platform,
+ snd_device_t out_snd_device,
+ snd_device_t in_snd_device)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ int acdb_rx_id, acdb_tx_id;
+ int ret = 0;
+
+ 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);
+ }
+ }
+ return ret;
+}
+
int platform_switch_voice_call_device_post(void *platform,
snd_device_t out_snd_device,
snd_device_t in_snd_device)
@@ -1666,7 +1728,7 @@
kv_pairs = str_parms_to_str(reply);
ALOGV_IF(kv_pairs != NULL, "%s: exit: returns - %s", __func__, kv_pairs);
- free(reply);
+ free(kv_pairs);
}
/* Delay in Us */
@@ -1700,3 +1762,69 @@
else
return false;
}
+
+/* Read offload buffer size from a property.
+ * If value is not power of 2 round it to
+ * power of 2.
+ */
+uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info)
+{
+ char value[PROPERTY_VALUE_MAX] = {0};
+ uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ if((property_get("audio.offload.buffer.size.kb", value, "")) &&
+ atoi(value)) {
+ fragment_size = atoi(value) * 1024;
+ }
+
+ if (info != NULL && info->has_video && info->is_streaming) {
+ fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+ ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+ __func__, out->compr_config.fragment_size);
+ }
+
+ fragment_size = ALIGN( fragment_size, 1024);
+
+ if(fragment_size < MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
+ fragment_size = MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ else if(fragment_size > MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
+ fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ ALOGV("%s: fragment_size %d", __func__, fragment_size);
+ return fragment_size;
+}
+
+uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
+{
+ uint32_t fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
+ uint32_t bits_per_sample = 16;
+
+ if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
+ bits_per_sample = 32;
+ }
+
+ if (!info->has_video) {
+ fragment_size = MAX_PCM_OFFLOAD_FRAGMENT_SIZE;
+
+ } else if (info->has_video && info->is_streaming) {
+ fragment_size = (PCM_OFFLOAD_BUFFER_DURATION_FOR_AV_STREAMING
+ * info->sample_rate
+ * bits_per_sample
+ * popcount(info->channel_mask))/1000;
+
+ } else if (info->has_video) {
+ fragment_size = (PCM_OFFLOAD_BUFFER_DURATION_FOR_AV
+ * info->sample_rate
+ * bits_per_sample
+ * popcount(info->channel_mask))/1000;
+ }
+
+ fragment_size = ALIGN( fragment_size, 1024);
+
+ if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
+ fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
+ else if(fragment_size > MAX_PCM_OFFLOAD_FRAGMENT_SIZE)
+ fragment_size = MAX_PCM_OFFLOAD_FRAGMENT_SIZE;
+
+ ALOGV("%s: fragment_size %d", __func__, fragment_size);
+ return fragment_size;
+}
+
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 3ea068d..e80d3f1 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -168,8 +168,6 @@
#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 INCALL_MUSIC_UPLINK_PCM_DEVICE 1
#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
@@ -213,11 +211,20 @@
#define QCHAT_CALL_PCM_DEVICE 20
#endif
+#ifdef PLATFORM_MSM8x26
+#define HFP_SCO_RX 28
+#define HFP_ASM_RX_TX 29
+#else
+#define HFP_SCO_RX 23
+#define HFP_ASM_RX_TX 24
+#endif
+
#define LIB_CSD_CLIENT "libcsd-client.so"
/* CSD-CLIENT related functions */
typedef int (*init_t)();
typedef int (*deinit_t)();
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);
@@ -234,6 +241,7 @@
init_t init;
deinit_t deinit;
disable_device_t disable_device;
+ enable_device_config_t enable_device_config;
enable_device_t enable_device;
volume_t volume;
mic_mute_t mic_mute;
diff --git a/hal/platform_api.h b/hal/platform_api.h
index a5f5074..81291a2 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -31,6 +31,9 @@
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);
+int platform_switch_voice_call_enable_device_config(void *platform,
+ snd_device_t out_snd_device,
+ snd_device_t in_snd_device);
int platform_switch_voice_call_device_post(void *platform,
snd_device_t out_snd_device,
snd_device_t in_snd_device);
@@ -62,4 +65,8 @@
/* From platform_info_parser.c */
int platform_info_init(void);
+struct audio_offload_info_t;
+uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
+uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
+
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 4db46f6..47ac2c8 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -59,8 +59,9 @@
struct pcm *pcm_rx;
struct pcm *pcm_tx;
struct stream_out *out_stream;
- int ref_count;
- int out_stream_count;
+ uint32_t out_stream_count;
+ uint32_t in_stream_count;
+ uint32_t sample_rate;
};
#define MODE_IS127 0x2
@@ -78,13 +79,15 @@
#define AUDIO_PARAMETER_VALUE_VOIP_TRUE "true"
#define AUDIO_PARAMETER_KEY_VOIP_CHECK "voip_flag"
#define AUDIO_PARAMETER_KEY_VOIP_OUT_STREAM_COUNT "voip_out_stream_count"
+#define AUDIO_PARAMETER_KEY_VOIP_SAMPLE_RATE "voip_sample_rate"
static struct voip_data voip_data = {
.pcm_rx = NULL,
.pcm_tx = NULL,
.out_stream = NULL,
- .ref_count = 0,
- .out_stream_count = 0
+ .out_stream_count = 0,
+ .in_stream_count = 0,
+ .sample_rate = 0
};
static int voip_set_volume(struct audio_device *adev, int volume);
@@ -280,10 +283,10 @@
int i, ret = 0;
struct audio_usecase *uc_info;
- ALOGD("%s: enter, ref_count=%d", __func__, voip_data.ref_count);
- voip_data.ref_count--;
+ ALOGD("%s: enter, out_stream_count=%d, in_stream_count=%d",
+ __func__, voip_data.out_stream_count, voip_data.in_stream_count);
- if (!voip_data.ref_count) {
+ if (!voip_data.out_stream_count && !voip_data.in_stream_count) {
uc_info = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
if (uc_info == NULL) {
ALOGE("%s: Could not find the usecase (%d) in the list",
@@ -310,8 +313,10 @@
list_remove(&uc_info->list);
free(uc_info);
+ voip_data.sample_rate = 0;
} else
- ALOGV("%s: NO-OP because ref_count=%d", __func__, voip_data.ref_count);
+ ALOGV("%s: NO-OP because out_stream_count=%d, in_stream_count=%d",
+ __func__, voip_data.out_stream_count, voip_data.in_stream_count);
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
@@ -327,12 +332,15 @@
ALOGD("%s: enter", __func__);
uc_info = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
- if ((uc_info == NULL) && (voip_data.out_stream)) {
+ if (uc_info == NULL) {
ALOGV("%s: voip usecase is added to the list", __func__);
uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
uc_info->id = USECASE_COMPRESS_VOIP_CALL;
uc_info->type = VOIP_CALL;
- uc_info->stream.out = voip_data.out_stream;
+ if (voip_data.out_stream)
+ uc_info->stream.out = voip_data.out_stream;
+ else
+ uc_info->stream.out = adev->primary_output;
uc_info->in_snd_device = SND_DEVICE_NONE;
uc_info->out_snd_device = SND_DEVICE_NONE;
@@ -388,12 +396,15 @@
ALOGE("%s: error %d\n", __func__, ret);
goto error_start_voip;
}
- voip_data.ref_count = 0;
- }
- else
+ } else {
ALOGV("%s: voip usecase is already enabled", __func__);
+ if (voip_data.out_stream)
+ uc_info->stream.out = voip_data.out_stream;
+ else
+ uc_info->stream.out = adev->primary_output;
+ select_devices(adev, USECASE_COMPRESS_VOIP_CALL);
+ }
- voip_data.ref_count++;
return 0;
error_start_voip:
@@ -472,6 +483,13 @@
str_parms_add_int(reply, AUDIO_PARAMETER_KEY_VOIP_OUT_STREAM_COUNT,
voip_data.out_stream_count);
}
+
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_VOIP_SAMPLE_RATE,
+ value, sizeof(value));
+ if (ret >= 0) {
+ str_parms_add_int(reply, AUDIO_PARAMETER_KEY_VOIP_SAMPLE_RATE,
+ voip_data.sample_rate);
+ }
}
void voice_extn_compress_voip_out_get_parameters(struct stream_out *out,
@@ -576,9 +594,9 @@
ALOGD("%s: enter", __func__);
+ voip_data.out_stream_count--;
ret = voip_stop_call(adev);
voip_data.out_stream = NULL;
- voip_data.out_stream_count--;
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
@@ -600,7 +618,7 @@
voip_data.out_stream = out;
voip_data.out_stream_count++;
-
+ voip_data.sample_rate = out->sample_rate;
ret = voip_set_mode(out->dev, out->format);
ALOGV("%s: exit", __func__);
@@ -615,6 +633,7 @@
ALOGD("%s: enter", __func__);
+ voip_data.in_stream_count--;
status = voip_stop_call(adev);
ALOGV("%s: exit: status(%d)", __func__, status);
@@ -630,15 +649,25 @@
ALOGD("%s: enter", __func__);
+ if ((voip_data.sample_rate != 0) &&
+ (voip_data.sample_rate != in->config.rate)) {
+ ret = -ENOTSUP;
+ goto done;
+ } else {
+ voip_data.sample_rate = in->config.rate;
+ }
+
in->usecase = USECASE_COMPRESS_VOIP_CALL;
if (in->config.rate == 16000)
in->config = pcm_config_voip_wb;
else
in->config = pcm_config_voip_nb;
+ voip_data.in_stream_count++;
ret = voip_set_mode(in->dev, in->format);
- ALOGV("%s: exit", __func__);
+done:
+ ALOGV("%s: exit, ret=%d", __func__, ret);
return ret;
}
@@ -734,7 +763,8 @@
if (ret) {
if ((popcount(config->channel_mask) == 1) &&
(config->sample_rate == 8000 || config->sample_rate == 16000))
- ret = true;
+ ret = ((voip_data.sample_rate == 0) ? true:
+ (voip_data.sample_rate == config->sample_rate));
else
ret = false;
}
diff --git a/mm-audio/aenc-aac/Android.mk b/mm-audio/aenc-aac/Android.mk
index 8ab45b3..7cd804a 100644
--- a/mm-audio/aenc-aac/Android.mk
+++ b/mm-audio/aenc-aac/Android.mk
@@ -24,6 +24,8 @@
ifeq ($(call is-board-platform,mpq8092),true)
include $(AENC_AAC_PATH)/qdsp6/Android.mk
endif
-
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_AAC_PATH)/qdsp6/Android.mk
+endif
endif
diff --git a/mm-audio/aenc-amrnb/Android.mk b/mm-audio/aenc-amrnb/Android.mk
index 2601ede..79474c6 100644
--- a/mm-audio/aenc-amrnb/Android.mk
+++ b/mm-audio/aenc-amrnb/Android.mk
@@ -24,6 +24,8 @@
ifeq ($(call is-board-platform,mpq8092),true)
include $(AENC_AMR_PATH)/qdsp6/Android.mk
endif
-
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_AMR_PATH)/qdsp6/Android.mk
+endif
endif
diff --git a/mm-audio/aenc-evrc/Android.mk b/mm-audio/aenc-evrc/Android.mk
index 2f42d6b..649f860 100644
--- a/mm-audio/aenc-evrc/Android.mk
+++ b/mm-audio/aenc-evrc/Android.mk
@@ -24,6 +24,8 @@
ifeq ($(call is-board-platform,mpq8092),true)
include $(AENC_EVRC_PATH)/qdsp6/Android.mk
endif
-
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_EVRC_PATH)/qdsp6/Android.mk
+endif
endif
diff --git a/mm-audio/aenc-qcelp13/Android.mk b/mm-audio/aenc-qcelp13/Android.mk
index fe18efc..c919f19 100644
--- a/mm-audio/aenc-qcelp13/Android.mk
+++ b/mm-audio/aenc-qcelp13/Android.mk
@@ -24,5 +24,8 @@
ifeq ($(call is-board-platform,mpq8092),true)
include $(AENC_QCELP13_PATH)/qdsp6/Android.mk
endif
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_QCELP13_PATH)/qdsp6/Android.mk
+endif
endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 5142353..a774c9e 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -1116,16 +1116,6 @@
return false;
}
#endif
-
- // Check if offload has been disabled
- char propValue[PROPERTY_VALUE_MAX];
- if (property_get("audio.offload.disable", propValue, "0")) {
- if (atoi(propValue) != 0) {
- ALOGV("offload disabled by audio.offload.disable=%s", propValue );
- return false;
- }
- }
-
// Check if stream type is music, then only allow offload as of now.
if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
{
@@ -1133,26 +1123,65 @@
return false;
}
- //TODO: enable audio offloading with video when ready
- if (offloadInfo.has_video)
- {
- if(property_get("av.offload.enable", propValue, NULL)) {
+ char propValue[PROPERTY_VALUE_MAX];
+ bool pcmOffload = false;
+ if (audio_is_offload_pcm(offloadInfo.format)) {
+ if(property_get("audio.offload.pcm.enable", propValue, NULL)) {
bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
- if (!prop_enabled) {
- ALOGW("offload disabled by av.offload.enable = %s ", propValue );
- return false;
+ if (prop_enabled) {
+ ALOGW("PCM offload property is enabled");
+ pcmOffload = true;
}
}
- if(offloadInfo.is_streaming &&
- property_get("av.streaming.offload.enable", propValue, NULL)) {
- bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
- if (!prop_enabled) {
- ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
- return false;
+ if (!pcmOffload) {
+ ALOGV("PCM offload disabled by property audio.offload.pcm.enable");
+ return false;
+ }
+ }
+
+ if (!pcmOffload) {
+ // Check if offload has been disabled
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+ return false;
}
}
- ALOGV("isOffloadSupported: has_video == true, property\
- set to enable offload");
+
+ //check if it's multi-channel AAC format
+ if (AudioSystem::popCount(offloadInfo.channel_mask) > 2
+ && offloadInfo.format == AUDIO_FORMAT_AAC) {
+ ALOGV("offload disabled for multi-channel AAC format");
+ return false;
+ }
+
+ if (offloadInfo.has_video)
+ {
+ if(property_get("av.offload.enable", propValue, NULL)) {
+ bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ if (!prop_enabled) {
+ ALOGW("offload disabled by av.offload.enable = %s ", propValue );
+ return false;
+ }
+ } else {
+ return false;
+ }
+
+ if(offloadInfo.is_streaming) {
+ if (property_get("av.streaming.offload.enable", propValue, NULL)) {
+ bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ if (!prop_enabled) {
+ ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
+ return false;
+ }
+ } else {
+ //Do not offload AV streamnig if the property is not defined
+ return false;
+ }
+ }
+ ALOGV("isOffloadSupported: has_video == true, property\
+ set to enable offload");
+ }
}
//If duration is less than minimum value defined in property, return false
@@ -1165,7 +1194,7 @@
ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
//duration checks only valid for MP3/AAC formats,
//do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
- if (offloadInfo.format == AUDIO_FORMAT_MP3 || offloadInfo.format == AUDIO_FORMAT_AAC)
+ if (offloadInfo.format == AUDIO_FORMAT_MP3 || offloadInfo.format == AUDIO_FORMAT_AAC || pcmOffload)
return false;
}
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index b6966e6..2cb910c 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -25,7 +25,7 @@
LOCAL_C_INCLUDES := \
external/tinyalsa/include \
- kernel/include/sound \
+ $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include \
$(call include-path-for, audio-effects)
include $(BUILD_SHARED_LIBRARY)
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index c724b58..9e6c1fc 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -23,7 +23,7 @@
#include <cutils/list.h>
#include <cutils/log.h>
#include <tinyalsa/asoundlib.h>
-#include <audio_effects.h>
+#include <sound/audio_effects.h>
#include <audio_effects/effect_bassboost.h>
#include "effect_api.h"
@@ -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/bundle.h b/post_proc/bundle.h
index a8e0f93..7ebea92 100644
--- a/post_proc/bundle.h
+++ b/post_proc/bundle.h
@@ -21,7 +21,7 @@
#define OFFLOAD_EFFECT_BUNDLE_H
#include <tinyalsa/asoundlib.h>
-#include <audio_effects.h>
+#include <sound/audio_effects.h>
#include "effect_api.h"
/* Retry for delay for mixer open */
diff --git a/post_proc/effect_api.c b/post_proc/effect_api.c
index a2e4f45..0b77969 100644
--- a/post_proc/effect_api.c
+++ b/post_proc/effect_api.c
@@ -33,7 +33,7 @@
#include <stdbool.h>
#include <cutils/log.h>
#include <tinyalsa/asoundlib.h>
-#include <audio_effects.h>
+#include <sound/audio_effects.h>
#include "effect_api.h"
diff --git a/post_proc/equalizer.c b/post_proc/equalizer.c
index 7c7ced2..bde8ef8 100644
--- a/post_proc/equalizer.c
+++ b/post_proc/equalizer.c
@@ -23,7 +23,7 @@
#include <cutils/list.h>
#include <cutils/log.h>
#include <tinyalsa/asoundlib.h>
-#include <audio_effects.h>
+#include <sound/audio_effects.h>
#include <audio_effects/effect_equalizer.h>
#include "effect_api.h"
diff --git a/post_proc/reverb.c b/post_proc/reverb.c
index d104073..4c7fe25 100644
--- a/post_proc/reverb.c
+++ b/post_proc/reverb.c
@@ -23,7 +23,7 @@
#include <cutils/list.h>
#include <cutils/log.h>
#include <tinyalsa/asoundlib.h>
-#include <audio_effects.h>
+#include <sound/audio_effects.h>
#include <audio_effects/effect_environmentalreverb.h>
#include <audio_effects/effect_presetreverb.h>
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index e9eb728..4190129 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -23,7 +23,7 @@
#include <cutils/list.h>
#include <cutils/log.h>
#include <tinyalsa/asoundlib.h>
-#include <audio_effects.h>
+#include <sound/audio_effects.h>
#include <audio_effects/effect_virtualizer.h>
#include "effect_api.h"
@@ -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;
}