Merge "hal: Fix input device selection for stereo recording"
diff --git a/hal/Android.mk b/hal/Android.mk
index 037be56..a7e0a02 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -26,6 +26,7 @@
LOCAL_SRC_FILES := \
audio_hw.c \
voice.c \
+ platform_info.c \
$(AUDIO_PLATFORM)/platform.c
LOCAL_SRC_FILES += audio_extn/audio_extn.c
@@ -97,19 +98,30 @@
LOCAL_CFLAGS += -DDS1_DOLBY_DDP_ENABLED
LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
+ LOCAL_SRC_FILES += audio_extn/dolby.c
endif
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_DS1_DOLBY_DAP)),true)
+ LOCAL_CFLAGS += -DDS1_DOLBY_DAP_ENABLED
+ifeq ($(strip $(AUDIO_FEATURE_DISABLED_DS1_DOLBY_DDP)),true)
+ LOCAL_SRC_FILES += audio_extn/dolby.c
+endif
+endif
+
+
LOCAL_SHARED_LIBRARIES := \
liblog \
libcutils \
libtinyalsa \
libtinycompress \
libaudioroute \
- libdl
+ libdl \
+ libexpat
LOCAL_C_INCLUDES += \
external/tinyalsa/include \
external/tinycompress/include \
+ external/expat/lib \
$(call include-path-for, audio-route) \
$(call include-path-for, audio-effects) \
$(LOCAL_PATH)/$(AUDIO_PLATFORM) \
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 80ce063..89903ba 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -29,8 +29,6 @@
#include "audio_hw.h"
#include "audio_extn.h"
-#include "sound/compress_params.h"
-
#define MAX_SLEEP_RETRY 100
#define WIFI_INIT_WAIT_SLEEP 50
@@ -321,6 +319,7 @@
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);
}
void audio_extn_get_parameters(const struct audio_device *adev,
@@ -359,57 +358,3 @@
return 0;
}
#endif /* AUXPCM_BT_ENABLED */
-
-
-#ifdef DS1_DOLBY_DDP_ENABLED
-
-bool audio_extn_dolby_is_supported_format(audio_format_t format)
-{
- if (format == AUDIO_FORMAT_AC3 ||
- format == AUDIO_FORMAT_EAC3)
- return true;
- else
- return false;
-}
-
-int audio_extn_dolby_get_snd_codec_id(audio_format_t format)
-{
- int id = 0;
-
- switch (format) {
- case AUDIO_FORMAT_AC3:
- id = SND_AUDIOCODEC_AC3;
- break;
- case AUDIO_FORMAT_EAC3:
- id = SND_AUDIOCODEC_EAC3;
- break;
- default:
- ALOGE("%s: Unsupported audio format :%x", __func__, format);
- }
-
- return id;
-}
-
-int audio_extn_dolby_set_DMID(struct audio_device *adev)
-{
- struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "DS1 Security";
- char c_dmid[128] = {0};
- int i_dmid, ret;
-
- property_get("dmid",c_dmid,"0");
- i_dmid = atoi(c_dmid);
-
- ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
- if (!ctl) {
- ALOGE("%s: Could not get ctl for mixer cmd - %s",
- __func__, mixer_ctl_name);
- return -EINVAL;
- }
- ALOGV("%s Dolby device manufacturer id is:%d",__func__,i_dmid);
- ret = mixer_ctl_set_value(ctl, 0, i_dmid);
-
- return ret;
-}
-#endif /* DS1_DOLBY_DDP_ENABLED */
-
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 72f8642..fb428db 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -155,14 +155,31 @@
void audio_extn_compr_cap_deinit();
#endif
-#ifndef DS1_DOLBY_DDP_ENABLED
-#define audio_extn_dolby_is_supported_format(format) (0)
-#define audio_extn_dolby_get_snd_codec_id(format) (0)
-#define audio_extn_dolby_set_DMID(adev) (0)
+#if defined(DS1_DOLBY_DDP_ENABLED) || defined(DS1_DOLBY_DAP_ENABLED)
+void audio_extn_dolby_set_dmid(struct audio_device *adev);
#else
-bool audio_extn_dolby_is_supported_format(audio_format_t format);
-int audio_extn_dolby_get_snd_codec_id(audio_format_t format);
-int audio_extn_dolby_set_DMID(struct audio_device *adev);
+#define audio_extn_dolby_set_dmid(adev) (0)
+#endif
+
+#ifndef DS1_DOLBY_DDP_ENABLED
+#define audio_extn_dolby_set_endpoint() (0)
+#else
+void audio_extn_dolby_set_endpoint(struct audio_device *adev);
+#endif
+
+#ifndef DS1_DOLBY_DDP_ENABLED
+#define audio_extn_ddp_set_parameters(adev, parms) (0)
+#define audio_extn_is_dolby_format(format) (0)
+#define audio_extn_dolby_get_snd_codec_id(format) (0)
+#define audio_extn_dolby_send_ddp_endp_params(adev) (0)
+#else
+bool audio_extn_is_dolby_format(audio_format_t format);
+int audio_extn_dolby_get_snd_codec_id(struct audio_device *adev,
+ struct stream_out *out,
+ audio_format_t format);
+void audio_extn_ddp_set_parameters(struct audio_device *adev,
+ struct str_parms *parms);
+void audio_extn_dolby_send_ddp_endp_params(struct audio_device *adev);
#endif
#ifndef HFP_ENABLED
diff --git a/hal/audio_extn/dolby.c b/hal/audio_extn/dolby.c
new file mode 100644
index 0000000..bcc7381
--- /dev/null
+++ b/hal/audio_extn/dolby.c
@@ -0,0 +1,456 @@
+/*
+ * Copyright (c) 2011-2014, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_dolby"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <cutils/properties.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include "audio_extn.h"
+#include "sound/compress_params.h"
+
+#ifdef DS1_DOLBY_DDP_ENABLED
+
+#define AUDIO_PARAMETER_DDP_DEV "ddp_device"
+#define AUDIO_PARAMETER_DDP_CH_CAP "ddp_chancap"
+#define AUDIO_PARAMETER_DDP_MAX_OUT_CHAN "ddp_maxoutchan"
+#define AUDIO_PARAMETER_DDP_OUT_MODE "ddp_outmode"
+#define AUDIO_PARAMETER_DDP_OUT_LFE_ON "ddp_outlfeon"
+#define AUDIO_PARAMETER_DDP_COMP_MODE "ddp_compmode"
+#define AUDIO_PARAMETER_DDP_STEREO_MODE "ddp_stereomode"
+
+#define PARAM_ID_MAX_OUTPUT_CHANNELS 0x00010DE2
+#define PARAM_ID_CTL_RUNNING_MODE 0x0
+#define PARAM_ID_CTL_ERROR_CONCEAL 0x00010DE3
+#define PARAM_ID_CTL_ERROR_MAX_RPTS 0x00010DE4
+#define PARAM_ID_CNV_ERROR_CONCEAL 0x00010DE5
+#define PARAM_ID_CTL_SUBSTREAM_SELECT 0x00010DE6
+#define PARAM_ID_CTL_INPUT_MODE 0x0
+#define PARAM_ID_OUT_CTL_OUTMODE 0x00010DE0
+#define PARAM_ID_OUT_CTL_OUTLFE_ON 0x00010DE1
+#define PARAM_ID_OUT_CTL_COMPMODE 0x00010D74
+#define PARAM_ID_OUT_CTL_STEREO_MODE 0x00010D76
+#define PARAM_ID_OUT_CTL_DUAL_MODE 0x00010D75
+#define PARAM_ID_OUT_CTL_DRCSCALE_HIGH 0x00010D7A
+#define PARAM_ID_OUT_CTL_DRCSCALE_LOW 0x00010D79
+#define PARAM_ID_OUT_CTL_OUT_PCMSCALE 0x00010D78
+#define PARAM_ID_OUT_CTL_MDCT_BANDLIMIT 0x00010DE7
+#define PARAM_ID_OUT_CTL_DRC_SUPPRESS 0x00010DE8
+
+/* DS1-DDP Endp Params */
+#define DDP_ENDP_NUM_PARAMS 17
+#define DDP_ENDP_NUM_DEVICES 22
+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,
+ PARAM_ID_CNV_ERROR_CONCEAL, PARAM_ID_CTL_SUBSTREAM_SELECT,
+ PARAM_ID_CTL_INPUT_MODE, PARAM_ID_OUT_CTL_OUTMODE,
+ PARAM_ID_OUT_CTL_OUTLFE_ON, PARAM_ID_OUT_CTL_COMPMODE,
+ PARAM_ID_OUT_CTL_STEREO_MODE, PARAM_ID_OUT_CTL_DUAL_MODE,
+ PARAM_ID_OUT_CTL_DRCSCALE_HIGH, PARAM_ID_OUT_CTL_DRCSCALE_LOW,
+ PARAM_ID_OUT_CTL_OUT_PCMSCALE, PARAM_ID_OUT_CTL_MDCT_BANDLIMIT,
+ PARAM_ID_OUT_CTL_DRC_SUPPRESS
+};
+
+static struct ddp_endp_params {
+ int device;
+ int dev_ch_cap;
+ int param_val[DDP_ENDP_NUM_PARAMS];
+ bool is_param_valid[DDP_ENDP_NUM_PARAMS];
+} ddp_endp_params[DDP_ENDP_NUM_DEVICES] = {
+ {AUDIO_DEVICE_OUT_EARPIECE, 2,
+ {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_SPEAKER, 2,
+ {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_WIRED_HEADSET, 2,
+ {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_WIRED_HEADPHONE, 2,
+ {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_BLUETOOTH_SCO, 2,
+ {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_BLUETOOTH_SCO_HEADSET, 2,
+ {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_BLUETOOTH_SCO_CARKIT, 2,
+ {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_BLUETOOTH_A2DP, 2,
+ {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_BLUETOOTH_A2DP_HEADPHONES, 2,
+ {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_BLUETOOTH_A2DP_SPEAKER, 2,
+ {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_AUX_DIGITAL, 2,
+ {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_AUX_DIGITAL, 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} },
+ {AUDIO_DEVICE_OUT_AUX_DIGITAL, 8,
+ {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_ANLG_DOCK_HEADSET, 2,
+ {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_DGTL_DOCK_HEADSET, 2,
+ {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_USB_ACCESSORY, 2,
+ {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_USB_DEVICE, 2,
+ {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_FM, 2,
+ {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_FM_TX, 2,
+ {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_ANC_HEADSET, 2,
+ {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_ANC_HEADPHONE, 2,
+ {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},
+ {1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
+};
+
+int update_ddp_endp_table(int device, int dev_ch_cap, int param_id,
+ int param_val)
+{
+ int idx = 0;
+ int param_idx = 0;
+ ALOGV("%s: dev 0x%x dev_ch_cap %d param_id 0x%x param_val %d",
+ __func__, device, dev_ch_cap , param_id, param_val);
+
+ for(idx=0; idx<DDP_ENDP_NUM_DEVICES; idx++) {
+ if(ddp_endp_params[idx].device == device) {
+ if(ddp_endp_params[idx].dev_ch_cap == dev_ch_cap) {
+ break;
+ }
+ }
+ }
+
+ if(idx>=DDP_ENDP_NUM_DEVICES) {
+ ALOGE("%s: device not available in DDP endp config table", __func__);
+ return -EINVAL;
+ }
+
+ for(param_idx=0; param_idx<DDP_ENDP_NUM_PARAMS; param_idx++) {
+ if (ddp_endp_params_id[param_idx] == param_id) {
+ break;
+ }
+ }
+
+ if(param_idx>=DDP_ENDP_NUM_PARAMS) {
+ ALOGE("param not available in DDP endp config table");
+ return -EINVAL;
+ }
+
+ ALOGV("ddp_endp_params[%d].param_val[%d] = %d", idx, param_idx, param_val);
+ ddp_endp_params[idx].param_val[param_idx] = param_val;
+ return 0;
+}
+
+void send_ddp_endp_params_stream(struct stream_out *out,
+ int device, int dev_ch_cap,
+ bool set_cache)
+{
+ int idx, i;
+ int ddp_endp_params_data[2*DDP_ENDP_NUM_PARAMS + 1];
+ int length = 0;
+ for(idx=0; idx<DDP_ENDP_NUM_DEVICES; idx++) {
+ if(ddp_endp_params[idx].device & device) {
+ if(ddp_endp_params[idx].dev_ch_cap == dev_ch_cap) {
+ break;
+ }
+ }
+ }
+ if(idx>=DDP_ENDP_NUM_DEVICES) {
+ ALOGE("device not available in DDP endp config table");
+ return;
+ }
+
+ length += 1; /* offset 0 is for num of parameter. increase offset by 1 */
+ for (i=0; i<DDP_ENDP_NUM_PARAMS; i++) {
+ if(ddp_endp_params[idx].is_param_valid[i]) {
+ ddp_endp_params_data[length++] = ddp_endp_params_id[i];
+ ddp_endp_params_data[length++] = ddp_endp_params[idx].param_val[i];
+ }
+ }
+ ddp_endp_params_data[0] = (length-1)/2;
+ if(length) {
+ char mixer_ctl_name[128];
+ struct audio_device *adev = out->dev;
+ struct mixer_ctl *ctl;
+ int pcm_device_id = platform_get_pcm_device_id(out->usecase,
+ PCM_PLAYBACK);
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Audio Stream %d Dec Params", pcm_device_id);
+ 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;
+ }
+ mixer_ctl_set_array(ctl, ddp_endp_params_data, length);
+ }
+ return;
+}
+
+void send_ddp_endp_params(struct audio_device *adev,
+ int ddp_dev, int dev_ch_cap)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if ((usecase->type == PCM_PLAYBACK) &&
+ (usecase->devices & ddp_dev) &&
+ (usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ ((usecase->stream.out->format == AUDIO_FORMAT_AC3) ||
+ (usecase->stream.out->format == AUDIO_FORMAT_EAC3))) {
+ send_ddp_endp_params_stream(usecase->stream.out, ddp_dev,
+ dev_ch_cap, false /* set cache */);
+ }
+ }
+}
+
+void audio_extn_dolby_send_ddp_endp_params(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if ((usecase->type == PCM_PLAYBACK) &&
+ (usecase->devices & AUDIO_DEVICE_OUT_ALL) &&
+ (usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ ((usecase->stream.out->format == AUDIO_FORMAT_AC3) ||
+ (usecase->stream.out->format == AUDIO_FORMAT_EAC3))) {
+ 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 */);
+ }
+ }
+}
+
+void audio_extn_ddp_set_parameters(struct audio_device *adev,
+ struct str_parms *parms)
+{
+ int ddp_dev, dev_ch_cap;
+ int val, ret;
+ char value[32]={0};
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_DEV, value,
+ sizeof(value));
+ if (ret >= 0) {
+ ddp_dev = atoi(value);
+ if (!(AUDIO_DEVICE_OUT_ALL & ddp_dev))
+ return;
+ } else
+ return;
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_CH_CAP, value,
+ sizeof(value));
+ if (ret >= 0) {
+ dev_ch_cap = atoi(value);
+ if ((dev_ch_cap != 2) && (dev_ch_cap != 6) && (dev_ch_cap != 8))
+ return;
+ } else
+ return;
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_MAX_OUT_CHAN, value,
+ sizeof(value));
+ if (ret >= 0) {
+ val = atoi(value);
+ update_ddp_endp_table(ddp_dev, dev_ch_cap,
+ PARAM_ID_MAX_OUTPUT_CHANNELS, val);
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_OUT_MODE, value,
+ sizeof(value));
+ if (ret >= 0) {
+ val = atoi(value);
+ update_ddp_endp_table(ddp_dev, dev_ch_cap,
+ PARAM_ID_OUT_CTL_OUTMODE, val);
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_OUT_LFE_ON, value,
+ sizeof(value));
+ if (ret >= 0) {
+ val = atoi(value);
+ update_ddp_endp_table(ddp_dev, dev_ch_cap,
+ PARAM_ID_OUT_CTL_OUTLFE_ON, val);
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_COMP_MODE, value,
+ sizeof(value));
+ if (ret >= 0) {
+ val = atoi(value);
+ update_ddp_endp_table(ddp_dev, dev_ch_cap,
+ PARAM_ID_OUT_CTL_COMPMODE, val);
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DDP_STEREO_MODE, value,
+ sizeof(value));
+ if (ret >= 0) {
+ val = atoi(value);
+ update_ddp_endp_table(ddp_dev, dev_ch_cap,
+ PARAM_ID_OUT_CTL_STEREO_MODE, val);
+ }
+
+ send_ddp_endp_params(adev, ddp_dev, dev_ch_cap);
+}
+
+int audio_extn_dolby_get_snd_codec_id(struct audio_device *adev,
+ struct stream_out *out,
+ audio_format_t format)
+{
+ int id = 0;
+
+ 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 */);
+#ifndef DS1_DOLBY_DAP_ENABLED
+ audio_extn_dolby_set_dmid(adev);
+#endif
+ break;
+ 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 */);
+#ifndef DS1_DOLBY_DAP_ENABLED
+ audio_extn_dolby_set_dmid(adev);
+#endif
+ break;
+ default:
+ ALOGE("%s: Unsupported audio format :%x", __func__, format);
+ }
+
+ return id;
+}
+
+bool audio_extn_is_dolby_format(audio_format_t format)
+{
+ if (format == AUDIO_FORMAT_AC3 ||
+ format == AUDIO_FORMAT_EAC3)
+ return true;
+ else
+ return false;
+}
+
+#endif /* DS1_DOLBY_DDP_ENABLED */
+
+#ifdef DS1_DOLBY_DAP_ENABLED
+void audio_extn_dolby_set_endpoint(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ struct mixer_ctl *ctl;
+ const char *mixer_ctl_name = "DS1 DAP Endpoint";
+ int endpoint = 0, ret;
+ bool send = false;
+
+ 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)) {
+ endpoint |= usecase->devices & AUDIO_DEVICE_OUT_ALL;
+ send = true;
+ }
+ }
+ if (!send)
+ 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;
+ }
+ ret = mixer_ctl_set_value(ctl, 0, endpoint);
+ if (ret)
+ ALOGE("%s: Dolby set endpint cannot be set error:%d",__func__, ret);
+
+ return;
+}
+#endif /* DS1_DOLBY_DAP_ENABLED */
+
+
+#if defined(DS1_DOLBY_DDP_ENABLED) || defined(DS1_DOLBY_DAP_ENABLED)
+void audio_extn_dolby_set_dmid(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ struct mixer_ctl *ctl;
+ const char *mixer_ctl_name = "DS1 Security";
+ char c_dmid[128] = {0};
+ int i_dmid, ret;
+ bool send = false;
+
+ 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))
+ send = true;
+ }
+ if (!send)
+ return;
+
+ property_get("dmid",c_dmid,"0");
+ i_dmid = atoi(c_dmid);
+
+ 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;
+ }
+ ALOGV("%s Dolby device manufacturer id is:%d",__func__,i_dmid);
+ ret = mixer_ctl_set_value(ctl, 0, i_dmid);
+ if (ret)
+ ALOGE("%s: Dolby DMID cannot be set error:%d",__func__, ret);
+
+ return;
+}
+#endif /* DS1_DOLBY_DDP_ENABLED || DS1_DOLBY_DAP_ENABLED */
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 19fdda2..04a7aba 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -53,8 +53,10 @@
#include "voice_extn.h"
#include "sound/compress_params.h"
+
#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
-#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (8 * 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 */
@@ -152,8 +154,62 @@
static unsigned int audio_device_ref_count;
static int set_voice_volume_l(struct audio_device *adev, float volume);
-static uint32_t get_offload_buffer_size();
-static int set_gapless_mode(struct audio_device *adev);
+
+/* 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) {
+
+
+ char value[PROPERTY_VALUE_MAX] = {0};
+ bool gapless_enabled = false;
+ const char *mixer_ctl_name = "Compress Gapless Playback";
+ struct mixer_ctl *ctl;
+
+ ALOGV("%s:", __func__);
+ property_get("audio.offload.gapless.enabled", value, NULL);
+ gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
+ ALOGE("%s: Could not set gapless mode %d",
+ __func__, gapless_enabled);
+ return -EINVAL;
+ }
+ return 0;
+}
static bool is_supported_format(audio_format_t format)
{
@@ -199,6 +255,10 @@
else
snd_device = usecase->out_snd_device;
+#ifdef DS1_DOLBY_DAP_ENABLED
+ audio_extn_dolby_set_dmid(adev);
+ audio_extn_dolby_set_endpoint(adev);
+#endif
strcpy(mixer_path, use_case_table[usecase->id]);
platform_add_backend_name(mixer_path, snd_device);
ALOGV("%s: apply mixer path: %s", __func__, mixer_path);
@@ -831,6 +891,7 @@
{
struct stream_out *out = (struct stream_out *) context;
struct listnode *item;
+ int ret = 0;
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
set_sched_policy(0, SP_FOREGROUND);
@@ -880,8 +941,14 @@
event = STREAM_CBK_EVENT_WRITE_READY;
break;
case OFFLOAD_CMD_PARTIAL_DRAIN:
- compress_next_track(out->compr);
- compress_partial_drain(out->compr);
+ ret = compress_next_track(out->compr);
+ if(ret == 0)
+ compress_partial_drain(out->compr);
+ 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;
@@ -1116,6 +1183,11 @@
if (out->offload_callback)
compress_nonblock(out->compr, out->non_blocking);
+#ifdef DS1_DOLBY_DDP_ENABLED
+ if (audio_extn_is_dolby_format(out->format))
+ audio_extn_dolby_send_ddp_endp_params(adev);
+#endif
+
if (adev->visualizer_start_output != NULL)
adev->visualizer_start_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_start_output != NULL)
@@ -1284,11 +1356,21 @@
struct compr_gapless_mdata tmp_mdata;
tmp_mdata.encoder_delay = 0;
tmp_mdata.encoder_padding = 0;
+
if (!out || !parms) {
ALOGE("%s: return invalid ",__func__);
return -EINVAL;
}
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FORMAT, value, sizeof(value));
+ if (ret >= 0) {
+ if (atoi(value) == SND_AUDIOSTREAMFORMAT_MP4ADTS) {
+ out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
+ ALOGV("ADTS format is set in offload mode");
+ }
+ out->send_new_metadata = 1;
+ }
+
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
if(ret >= 0)
is_meta_data_params = true;
@@ -1449,7 +1531,8 @@
voice_extn_out_get_parameters(out, query, reply);
str = str_parms_to_str(reply);
if (!strncmp(str, "", sizeof(""))) {
- str = strdup(keys);
+ free(str);
+ str = strdup(keys);
}
}
str_parms_destroy(query);
@@ -1826,6 +1909,18 @@
/* no audio source uses val == 0 */
if ((in->source != val) && (val != 0)) {
in->source = val;
+ if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+ (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ (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) {
+ ALOGE("%s: Compress voip input cannot be opened, error:%d",
+ __func__, ret);
+ goto done;
+ }
+ }
}
}
@@ -1840,6 +1935,7 @@
}
}
+done:
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
@@ -2053,7 +2149,7 @@
goto error_open;
}
if (!is_supported_format(config->offload_info.format) &&
- !audio_extn_dolby_is_supported_format(config->offload_info.format)) {
+ !audio_extn_is_dolby_format(config->offload_info.format)) {
ALOGE("%s: Unsupported audio format", __func__);
ret = -EINVAL;
goto error_open;
@@ -2076,13 +2172,14 @@
out->stream.drain = out_drain;
out->stream.flush = out_flush;
- if (audio_extn_dolby_is_supported_format(config->offload_info.format))
+ if (audio_extn_is_dolby_format(config->offload_info.format))
out->compr_config.codec->id =
- audio_extn_dolby_get_snd_codec_id(config->offload_info.format);
+ audio_extn_dolby_get_snd_codec_id(adev, out,
+ config->offload_info.format);
else
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
- out->compr_config.fragment_size = get_offload_buffer_size();
+ out->compr_config.fragment_size = get_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);
@@ -2091,6 +2188,7 @@
out->compr_config.codec->ch_in =
popcount(config->channel_mask);
out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
+ out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
@@ -2103,18 +2201,8 @@
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
-
- if (audio_extn_dolby_is_supported_format(out->format)) {
- ret = audio_extn_dolby_set_DMID(adev);
- if (ret != 0) {
- ALOGE("%s: Dolby DMID cannot be set error:%d",
- __func__, ret);
- goto error_open;
- }
- }
-
//Decide if we need to use gapless mode by default
- set_gapless_mode(adev);
+ 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);
@@ -2448,16 +2536,7 @@
in->config.rate = config->sample_rate;
in->format = config->format;
- if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
- (voice_extn_compress_voip_is_config_supported(config))) {
- ret = voice_extn_compress_voip_open_input_stream(in);
- if (ret != 0)
- {
- ALOGE("%s: Compress voip input cannot be opened, error:%d",
- __func__, ret);
- goto err_open;
- }
- } else if (channel_count == 6) {
+ if (channel_count == 6) {
if(audio_extn_ssr_get_enabled()) {
if(audio_extn_ssr_init(adev, in)) {
ALOGE("%s: audio_extn_ssr_init failed", __func__);
@@ -2469,7 +2548,8 @@
goto err_open;
}
} else if (audio_extn_compr_cap_enabled() &&
- audio_extn_compr_cap_format_supported(config->format)) {
+ audio_extn_compr_cap_format_supported(config->format) &&
+ (in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
audio_extn_compr_cap_init(adev, in);
} else {
in->config.channels = channel_count;
@@ -2650,55 +2730,6 @@
return 0;
}
-/* 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()
-{
- 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(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 set_gapless_mode(struct audio_device *adev) {
-
-
- char value[PROPERTY_VALUE_MAX] = {0};
- bool gapless_enabled = false;
- const char *mixer_ctl_name = "Compress Gapless Playback";
- struct mixer_ctl *ctl;
-
- ALOGV("%s:", __func__);
- property_get("audio.offload.gapless.enabled", value, NULL);
- gapless_enabled = atoi(value) || !strncmp("true", value, 4);
-
- ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
- if (!ctl) {
- ALOGE("%s: Could not get ctl for mixer cmd - %s",
- __func__, mixer_ctl_name);
- return -EINVAL;
- }
-
- if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
- ALOGE("%s: Could not set gapless mode %d",
- __func__, gapless_enabled);
- return -EINVAL;
- }
- return 0;
-
-}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 298c60d..c1ba595 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -391,6 +391,16 @@
return device_id;
}
+int platform_get_snd_device_index(char *snd_device_index_name)
+{
+ return -ENODEV;
+}
+
+int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
+{
+ return -ENODEV;
+}
+
int platform_send_audio_calibration(void *platform, snd_device_t snd_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 312c327..40fed4e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -76,7 +76,7 @@
/* Audio calibration related functions */
typedef void (*acdb_deallocate_t)();
-typedef int (*acdb_init_t)();
+typedef int (*acdb_init_t)(char *);
typedef void (*acdb_send_audio_cal_t)(int, int);
typedef void (*acdb_send_voice_cal_t)(int, int);
typedef int (*acdb_reload_vocvoltable_t)(int);
@@ -222,7 +222,7 @@
};
/* ACDB IDs (audio DSP path configuration IDs) for each sound device */
-static const int acdb_device_table[SND_DEVICE_MAX] = {
+static int acdb_device_table[SND_DEVICE_MAX] = {
[SND_DEVICE_NONE] = -1,
[SND_DEVICE_OUT_HANDSET] = 7,
[SND_DEVICE_OUT_SPEAKER] = 14,
@@ -240,7 +240,7 @@
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
[SND_DEVICE_OUT_AFE_PROXY] = 0,
- [SND_DEVICE_OUT_USB_HEADSET] = 0,
+ [SND_DEVICE_OUT_USB_HEADSET] = 45,
[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 14,
[SND_DEVICE_OUT_TRANSMISSION_FM] = 0,
[SND_DEVICE_OUT_ANC_HEADSET] = 26,
@@ -294,6 +294,84 @@
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
};
+struct snd_device_index {
+ char name[100];
+ unsigned int index;
+};
+
+#define TO_NAME_INDEX(X) #X, X
+
+/* Used to get index from parsed sting */
+struct snd_device_index snd_device_name_index[SND_DEVICE_MAX] = {
+ {TO_NAME_INDEX(SND_DEVICE_OUT_HANDSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_AFE_PROXY)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_USB_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_TRANSMISSION_FM)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_FB_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_ANC_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AEC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AEC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_AEC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_AEC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HEADSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HEADSET_MIC_FLUENCE)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_HEADSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HDMI_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_DMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_QMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_MIC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_STEREO)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_USB_HEADSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_FM)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_AANC_HANDSET_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_QUAD_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+};
+
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
@@ -565,13 +643,16 @@
__func__, LIB_ACDB_LOADER);
my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
- "acdb_loader_init_ACDB");
+ "acdb_loader_init_v2");
if (my_data->acdb_init == NULL)
- ALOGE("%s: dlsym error %s for acdb_loader_init_ACDB", __func__, dlerror());
+ ALOGE("%s: dlsym error %s for acdb_loader_init_v2", __func__, dlerror());
else
- my_data->acdb_init();
+ my_data->acdb_init(snd_card_name);
}
+ /* 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
@@ -669,6 +750,46 @@
return device_id;
}
+int platform_get_snd_device_index(char *snd_device_index_name)
+{
+ int ret = 0;
+ int i;
+
+ if (snd_device_index_name == NULL) {
+ ALOGE("%s: snd_device_index_name is NULL", __func__);
+ ret = -ENODEV;
+ goto done;
+ }
+
+ for (i=0; i < SND_DEVICE_MAX; i++) {
+ if(strcmp(snd_device_name_index[i].name, snd_device_index_name) == 0) {
+ ret = snd_device_name_index[i].index;
+ goto done;
+ }
+ }
+ ALOGE("%s: Could not find index for snd_device_index_name = %s",
+ __func__, snd_device_index_name);
+ ret = -ENODEV;
+done:
+ return ret;
+}
+
+int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
+{
+ int ret = 0;
+
+ if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
+ ALOGE("%s: Invalid snd_device = %d",
+ __func__, snd_device);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ acdb_device_table[snd_device] = acdb_id;
+done:
+ return ret;
+}
+
int platform_send_audio_calibration(void *platform, snd_device_t snd_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index bb1f787..3ea068d 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -147,7 +147,7 @@
* the buffer size of an input/output stream
*/
#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
-#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 4
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
@@ -174,7 +174,7 @@
#define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
-#define SPKR_PROT_CALIB_TX_PCM_DEVICE 22
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 25
#define PLAYBACK_OFFLOAD_DEVICE 9
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 4096ef0..a5f5074 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -17,8 +17,8 @@
* limitations under the License.
*/
-#ifndef QCOM_AUDIO_PLATFORM_API_H
-#define QCOM_AUDIO_PLATFORM_API_H
+#ifndef AUDIO_PLATFORM_API_H
+#define AUDIO_PLATFORM_API_H
void *platform_init(struct audio_device *adev);
void platform_deinit(void *platform);
@@ -27,6 +27,8 @@
char *device_name);
void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
+int platform_get_snd_device_index(char *snd_device_index_name);
+int platform_set_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_device_post(void *platform,
@@ -57,4 +59,7 @@
bool platform_listen_update_status(snd_device_t snd_device);
-#endif // QCOM_AUDIO_PLATFORM_API_H
+/* From platform_info_parser.c */
+int platform_info_init(void);
+
+#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/platform_info.c b/hal/platform_info.c
new file mode 100644
index 0000000..8f56107
--- /dev/null
+++ b/hal/platform_info.c
@@ -0,0 +1,152 @@
+/*
+ * Copyright (c) 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:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "platform_info"
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <stdio.h>
+#include <expat.h>
+#include <cutils/log.h>
+#include <audio_hw.h>
+#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)
+{
+ int index;
+
+ if (strcmp(attr[0], "name") != 0) {
+ ALOGE("%s: 'name' not found, no ACDB ID set!", __func__);
+ goto done;
+ }
+
+ 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);
+ 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);
+ 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]));
+ goto done;
+ }
+
+done:
+ return;
+}
+
+static void start_tag(void *userdata, 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, "device") == 0)
+ process_device(attr);
+
+ return;
+}
+
+static void end_tag(void *userdata, const XML_Char *tag_name)
+{
+
+}
+
+int platform_info_init(void)
+{
+ XML_Parser parser;
+ FILE *file;
+ int ret = 0;
+ int bytes_read;
+ void *buf;
+
+ file = fopen(PLATFORM_INFO_XML_PATH, "r");
+ if (!file) {
+ ALOGD("%s: Failed to open %s, using defaults.",
+ __func__, PLATFORM_INFO_XML_PATH);
+ ret = -ENODEV;
+ goto done;
+ }
+
+ parser = XML_ParserCreate(NULL);
+ if (!parser) {
+ ALOGE("%s: Failed to create XML parser!", __func__);
+ ret = -ENODEV;
+ goto err_close_file;
+ }
+
+ XML_SetElementHandler(parser, start_tag, end_tag);
+
+ while (1) {
+ buf = XML_GetBuffer(parser, BUF_SIZE);
+ if (buf == NULL) {
+ ALOGE("%s: XML_GetBuffer failed", __func__);
+ ret = -ENOMEM;
+ goto err_free_parser;
+ }
+
+ bytes_read = fread(buf, 1, BUF_SIZE, file);
+ if (bytes_read < 0) {
+ ALOGE("%s: fread failed, bytes read = %d", __func__, bytes_read);
+ ret = bytes_read;
+ goto err_free_parser;
+ }
+
+ if (XML_ParseBuffer(parser, bytes_read,
+ bytes_read == 0) == XML_STATUS_ERROR) {
+ ALOGE("%s: XML_ParseBuffer failed, for %s",
+ __func__, PLATFORM_INFO_XML_PATH);
+ ret = -EINVAL;
+ goto err_free_parser;
+ }
+
+ if (bytes_read == 0)
+ break;
+ }
+
+err_free_parser:
+ XML_ParserFree(parser);
+err_close_file:
+ fclose(file);
+done:
+ return ret;
+}
diff --git a/hal_mpq/audio_stream_out.c b/hal_mpq/audio_stream_out.c
index c475a1e..454b1d6 100644
--- a/hal_mpq/audio_stream_out.c
+++ b/hal_mpq/audio_stream_out.c
@@ -1051,23 +1051,19 @@
}
/*TODO: do we need to apply volume at the session open*/
-static int set_compress_volume(struct alsa_handle *handle, int left, int right)
+static int set_compress_volume(struct alsa_handle *handle, float left, float right)
{
struct audio_device *adev = handle->out->dev;
struct mixer_ctl *ctl;
int volume[2];
- char mixer_ctl_name[44]; // max length of name is 44 as defined
- char device_id[STRING_LENGTH_OF_INTEGER+1];
+ char mixer_ctl_name[MIXER_PATH_MAX_LENGTH];
+ ALOGV("%s:setting volume l %f r %f ", __func__, left, right);
memset(mixer_ctl_name, 0, sizeof(mixer_ctl_name));
- strlcpy(mixer_ctl_name, "Compress Playback Volume", sizeof(mixer_ctl_name));
-
- memset(device_id, 0, sizeof(device_id));
- snprintf(device_id, "%d", handle->device_id, sizeof(device_id));
-
- strlcat(mixer_ctl_name, device_id, sizeof(mixer_ctl_name));
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Compress Playback %d Volume", handle->device_id);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -1075,8 +1071,8 @@
__func__, mixer_ctl_name);
return -EINVAL;
}
- volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX);
- volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX);
+ volume[0] = (int)(left * (float) COMPRESS_PLAYBACK_VOLUME_MAX);
+ volume[1] = (int)(right * (float) COMPRESS_PLAYBACK_VOLUME_MAX);
mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
return 0;
@@ -1709,8 +1705,8 @@
uc_info->out_snd_device = SND_DEVICE_NONE;
/* This must be called before adding this usecase to the list */
- //if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
- // check_and_set_hdmi_channels(adev, out->config.channels);
+ if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ check_and_set_hdmi_channels(adev, handle->config.channels);
list_add_tail(&adev->usecase_list, &uc_info->list);
@@ -2105,7 +2101,7 @@
struct alsa_handle *handle;
struct audio_device *adev = out->dev;
int ret = -ENOSYS;
- ALOGV("%s", __func__);
+ ALOGV("%s:setting volume l %f r %f ", __func__, left, right);
pthread_mutex_lock(&out->lock);
list_for_each(node, &out->session_list) {
handle = node_to_item(node, struct alsa_handle, list);
@@ -2118,7 +2114,7 @@
out->left_volume = left;
out->right_volume = right;
- //ret = set_compress_volume(handle, left, right);
+ ret = set_compress_volume(handle, left, right);
}
}
pthread_mutex_unlock(&out->lock);
diff --git a/hal_mpq/mpq8092/platform.c b/hal_mpq/mpq8092/platform.c
index 3c1b4f7..6c50034 100644
--- a/hal_mpq/mpq8092/platform.c
+++ b/hal_mpq/mpq8092/platform.c
@@ -176,6 +176,7 @@
[SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET] = "speaker-and-anc-headphones",
[SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
+ [SND_DEVICE_OUT_SPDIF] = "spdif",
/* Capture sound devices */
[SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -249,6 +250,7 @@
[SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET] = 26,
[SND_DEVICE_OUT_ANC_HANDSET] = 103,
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = 101,
+ [SND_DEVICE_OUT_SPDIF] = 18,
[SND_DEVICE_IN_HANDSET_MIC] = 4,
[SND_DEVICE_IN_HANDSET_MIC_AEC] = 106,
@@ -607,6 +609,8 @@
strlcat(mixer_path, " capture-fm", MIXER_PATH_MAX_LENGTH);
else if (snd_device == SND_DEVICE_OUT_TRANSMISSION_FM)
strlcat(mixer_path, " transmission-fm", MIXER_PATH_MAX_LENGTH);
+ else if (snd_device == SND_DEVICE_OUT_SPDIF)
+ strlcat(mixer_path, " spdif", MIXER_PATH_MAX_LENGTH);
}
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type)
@@ -897,6 +901,8 @@
snd_device = SND_DEVICE_OUT_TRANSMISSION_FM;
} else if (devices & AUDIO_DEVICE_OUT_EARPIECE) {
snd_device = SND_DEVICE_OUT_HANDSET;
+ } else if (devices & AUDIO_DEVICE_OUT_SPDIF) {
+ snd_device = SND_DEVICE_OUT_SPDIF;
} else {
ALOGE("%s: Unknown device(s) %#x", __func__, devices);
}
diff --git a/hal_mpq/mpq8092/platform.h b/hal_mpq/mpq8092/platform.h
index 2a81df5..562f979 100644
--- a/hal_mpq/mpq8092/platform.h
+++ b/hal_mpq/mpq8092/platform.h
@@ -74,6 +74,7 @@
SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
SND_DEVICE_OUT_ANC_HANDSET,
SND_DEVICE_OUT_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_SPDIF,
SND_DEVICE_OUT_END,
/*
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index b6a06e4..c68ab6e 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -30,6 +30,14 @@
LOCAL_CFLAGS += -DAUDIO_EXTN_INCALL_MUSIC_ENABLED
endif
+
+ifeq ($(strip $(TARGET_BOARD_PLATFORM)),msm8916)
+LOCAL_CFLAGS += -DVOICE_CONCURRENCY
+LOCAL_CFLAGS += -DWFD_CONCURRENCY
+endif
+
+
+
include $(BUILD_SHARED_LIBRARY)
endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index f64bbfe..5142353 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -418,12 +418,36 @@
AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(AudioSystem::stream_type stream)
{
-#ifdef QCOM_INCALL_MUSIC_ENABLED
- if (stream == AudioSystem::INCALL_MUSIC)
- return STRATEGY_MEDIA;
+ // stream to strategy mapping
+ switch (stream) {
+ case AudioSystem::VOICE_CALL:
+ case AudioSystem::BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AudioSystem::RING:
+ case AudioSystem::ALARM:
+ return STRATEGY_SONIFICATION;
+ case AudioSystem::NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AudioSystem::DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type");
+ case AudioSystem::SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AudioSystem::TTS:
+ case AudioSystem::MUSIC:
+#ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED
+ case AudioSystem::INCALL_MUSIC:
#endif
+#ifdef QCOM_INCALL_MUSIC_ENABLED
+ case AudioSystem::INCALL_MUSIC:
+#endif
+ return STRATEGY_MEDIA;
+ case AudioSystem::ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
+ }
- return getStrategy(stream);
}
audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
@@ -632,7 +656,8 @@
if (device2 == AUDIO_DEVICE_NONE) {
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
}
- if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+ && (device2 == AUDIO_DEVICE_NONE)) {
// no sonification on aux digital (e.g. HDMI)
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
}
@@ -642,12 +667,14 @@
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
}
#ifdef AUDIO_EXTN_FM_ENABLED
- if ((strategy != STRATEGY_SONIFICATION) && (device2 == AUDIO_DEVICE_NONE)) {
+ if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+ && (device2 == AUDIO_DEVICE_NONE)) {
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM_TX;
}
#endif
#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
- if ((strategy != STRATEGY_SONIFICATION) && (device2 == AUDIO_DEVICE_NONE)) {
+ if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+ && (device2 == AUDIO_DEVICE_NONE)) {
// no sonification on WFD sink
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY;
}
@@ -878,6 +905,422 @@
#endif
return AudioPolicyManagerBase::computeVolume(stream, index, output, device);
}
+
+
+audio_io_handle_t AudioPolicyManager::getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ AudioSystem::output_flags flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = 0;
+ uint32_t latency = 0;
+ routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ IOProfile *profile = NULL;
+
+#ifdef VOICE_CONCURRENCY
+ if (isInCall()) {
+ ALOGV(" IN call mode adding ULL flags .. flags: %x ", flags );
+ //For voip paths
+ if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+ flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+ else //route every thing else to ULL path
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ }
+#endif
+
+#ifdef WFD_CONCURRENCY
+ if ((mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY)
+ && (stream != AudioSystem::MUSIC)) {
+ ALOGV(" WFD mode adding ULL flags for non music stream.. flags: %x ", flags );
+ //For voip paths
+ if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+ flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+ else //route every thing else to ULL path
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ }
+#endif
+
+ ALOGV(" getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x ",
+ device, stream, samplingRate, format, channelMask, flags);
+
+
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mSamplingRate = mTestSamplingRate;
+ outputDesc->mFormat = mTestFormat;
+ outputDesc->mChannelMask = mTestChannels;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags,
+ offloadInfo);
+ if (mTestOutputs[mCurOutput]) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+
+ if ((format == AudioSystem::PCM_16_BIT) &&(AudioSystem::popCount(channelMask) > 2)) {
+ ALOGV("owerwrite flag(%x) for PCM16 multi-channel(CM:%x) playback", flags ,channelMask);
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if ((((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !isNonOffloadableEffectEnabled()) &&
+ flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != NULL) {
+ AudioOutputDescriptor *outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mId);
+ }
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mSamplingRate = samplingRate;
+ outputDesc->mFormat = (audio_format_t)format;
+ outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+ output = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags,
+ offloadInfo);
+
+ // only accept an output with the requested parameters
+ if (output == 0 ||
+ (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+ (format != 0 && format != outputDesc->mFormat) ||
+ (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != 0) {
+ mpClientInterface->closeOutput(output);
+ }
+ delete outputDesc;
+ return 0;
+ }
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ return output;
+ }
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm((audio_format_t)format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ output = selectOutput(outputs, flags);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV("getOutput() returns output %d", output);
+
+ return output;
+}
+
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGV(" isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%lld us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+#ifdef VOICE_CONCURRENCY
+ if(isInCall())
+ {
+ ALOGD("\n blocking compress offload on call mode\n");
+ 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)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ //TODO: enable audio offloading with video when ready
+ 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;
+ }
+ }
+ 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;
+ }
+ }
+ ALOGV("isOffloadSupported: has_video == true, property\
+ set to enable offload");
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ 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)
+ return false;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+ return (profile != NULL);
+}
+
+void AudioPolicyManager::setPhoneState(int state)
+
+{
+ ALOGV("setPhoneState() state %d", state);
+ audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+ if (state < 0 || state >= AudioSystem::NUM_MODES) {
+ ALOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ ALOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isInCall()) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, false, true);
+ }
+ }
+
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+ bool force = false;
+
+ // are we entering or starting a call
+ if (!isStateInCall(oldState) && isStateInCall(state)) {
+ ALOGV(" Entering call in setPhoneState()");
+ // force routing command to audio hardware when starting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+ }
+ } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+ ALOGV(" Exiting call in setPhoneState()");
+ // force routing command to audio hardware when exiting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+ }
+ } else if (isStateInCall(state) && (state != oldState)) {
+ ALOGV(" Switching between telephony and VoIP in setPhoneState()");
+ // force routing command to audio hardware when switching between telephony and VoIP
+ // even if no device change is needed
+ force = true;
+ }
+
+ // check for device and output changes triggered by new phone state
+ newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+ newDevice = hwOutputDesc->device();
+ }
+
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((desc->isStrategyActive(STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ desc->isStrategyActive(STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->mLatency*2)) {
+ delayMs = desc->mLatency*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ }
+
+ // change routing is necessary
+ setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, true, true);
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AudioSystem::MODE_RINGTONE &&
+ isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+
+#ifdef VOICE_CONCURRENCY
+ //Call invalidate to reset all opened non ULL audio tracks
+ if(isInCall())
+ {
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ ALOGV("\n Invalidate on call mode for stream :: %d \n", i);
+ //FIXME see fixme on name change
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+ 0 /* ignored */);
+ }
+ }
+#endif
+
+}
+
extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
{
return new AudioPolicyManager(clientInterface);
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 7a8cfa9..34ca701 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -48,6 +48,17 @@
uint32_t format,
uint32_t channels,
AudioSystem::audio_in_acoustics acoustics);
+ virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate = 0,
+ uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::output_flags flags =
+ AudioSystem::OUTPUT_FLAG_INDIRECT,
+ const audio_offload_info_t *offloadInfo = NULL);
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+ virtual void setPhoneState(int state);
protected:
// return the strategy corresponding to a given stream type
static routing_strategy getStrategy(AudioSystem::stream_type stream);
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index c64ba6b..c724b58 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -239,6 +239,11 @@
ALOGV("%s", __func__);
bass_ctxt->ctl = output->ctl;
ALOGV("output->ctl: %p", output->ctl);
+ if (offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass)))
+ if (bass_ctxt->ctl)
+ 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/equalizer.c b/post_proc/equalizer.c
index e31d2b9..7c7ced2 100644
--- a/post_proc/equalizer.c
+++ b/post_proc/equalizer.c
@@ -491,6 +491,11 @@
ALOGV("%s: %p", __func__, output->ctl);
eq_ctxt->ctl = output->ctl;
+ if (offload_eq_get_enable_flag(&(eq_ctxt->offload_eq)))
+ if (eq_ctxt->ctl)
+ offload_eq_send_params(eq_ctxt->ctl, eq_ctxt->offload_eq,
+ OFFLOAD_SEND_EQ_ENABLE_FLAG |
+ OFFLOAD_SEND_EQ_BANDS_LEVEL);
return 0;
}
diff --git a/post_proc/reverb.c b/post_proc/reverb.c
index 4fc8c83..d104073 100644
--- a/post_proc/reverb.c
+++ b/post_proc/reverb.c
@@ -236,9 +236,14 @@
void reverb_set_preset(reverb_context_t *context, int16_t preset)
{
+ bool enable;
ALOGV("%s: preset: %d", __func__, preset);
context->next_preset = preset;
offload_reverb_set_preset(&(context->offload_reverb), preset);
+
+ enable = (preset == REVERB_PRESET_NONE) ? false: true;
+ offload_reverb_set_enable_flag(&(context->offload_reverb), enable);
+
if (context->ctl)
offload_reverb_send_params(context->ctl, context->offload_reverb,
OFFLOAD_SEND_REVERB_ENABLE_FLAG |
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 2f0ca6b..e9eb728 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -237,6 +237,11 @@
ALOGV("%s", __func__);
virt_ctxt->ctl = output->ctl;
+ if (offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt)))
+ if (virt_ctxt->ctl)
+ offload_virtualizer_send_params(virt_ctxt->ctl, virt_ctxt->offload_virt,
+ OFFLOAD_SEND_VIRTUALIZER_ENABLE_FLAG |
+ OFFLOAD_SEND_VIRTUALIZER_STRENGTH);
return 0;
}