Merge "hal: add input source check to open compress voip input"
diff --git a/hal/Android.mk b/hal/Android.mk
index d25c926..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
@@ -114,23 +115,19 @@
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) \
$(LOCAL_PATH)/audio_extn \
$(LOCAL_PATH)/voice_extn
-ifneq ($(filter msm8974,$(AUDIO_PLATFORM)),)
- LOCAL_C_INCLUDES += external/expat/lib
- LOCAL_SHARED_LIBRARIES += libexpat
- LOCAL_SRC_FILES += $(AUDIO_PLATFORM)/platform_parser.c
-endif
-
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_LISTEN)),true)
LOCAL_CFLAGS += -DAUDIO_LISTEN_ENABLED
LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-listen
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index cfcc5a5..1a7b83e 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -152,8 +152,55 @@
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()
+{
+ 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 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)
{
@@ -835,6 +882,7 @@
{
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);
@@ -884,8 +932,14 @@
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;
@@ -2127,7 +2181,7 @@
__func__, config->offload_info.version,
config->offload_info.bit_rate);
//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);
@@ -2655,55 +2709,6 @@
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 54bda5a..c1ba595 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -391,6 +391,11 @@
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;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index b4d8c5d..a25991f 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -31,7 +31,6 @@
#include "platform.h"
#include "audio_extn.h"
#include "voice_extn.h"
-#include "platform_parser.h"
#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
#define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
@@ -295,6 +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)
@@ -673,6 +750,30 @@
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;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index bb1f787..ca8469a 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
diff --git a/hal/msm8974/platform_parser.h b/hal/msm8974/platform_parser.h
deleted file mode 100644
index 3e91934..0000000
--- a/hal/msm8974/platform_parser.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AUDIO_PLATFORM_PARSER_H
-#define AUDIO_PLATFORM_PARSER_H
-
-int platform_info_init(void);
-
-#endif // AUDIO_PLATFORM_PARSER_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0b2c435..a5f5074 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -27,6 +27,7 @@
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);
@@ -58,4 +59,7 @@
bool platform_listen_update_status(snd_device_t snd_device);
+/* From platform_info_parser.c */
+int platform_info_init(void);
+
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/msm8974/platform_parser.c b/hal/platform_info.c
similarity index 70%
rename from hal/msm8974/platform_parser.c
rename to hal/platform_info.c
index 8f86d97..8f56107 100644
--- a/hal/msm8974/platform_parser.c
+++ b/hal/platform_info.c
@@ -27,7 +27,7 @@
* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#define LOG_TAG "platform_parser"
+#define LOG_TAG "platform_info"
#define LOG_NDDEBUG 0
#include <errno.h>
@@ -35,41 +35,40 @@
#include <expat.h>
#include <cutils/log.h>
#include <audio_hw.h>
-#include <platform_api.h>
-#include "platform.h"
-#include "platform_parser.h"
+#include "platform_api.h"
+#include <platform.h>
-#define PLATFORM_XML_PATH "/system/etc/platform_info.xml"
+#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define BUF_SIZE 1024
-static void process_device(void *userdata, const XML_Char **attr)
+static void process_device(const XML_Char **attr)
{
- unsigned int *snd_device_index = userdata;
+ int index;
- if (strcmp(attr[0], "name") != 0)
+ if (strcmp(attr[0], "name") != 0) {
+ ALOGE("%s: 'name' not found, no ACDB ID set!", __func__);
goto done;
+ }
- if (platform_get_snd_device_name(*snd_device_index) == NULL)
- goto next;
- if (strcmp(attr[1], platform_get_snd_device_name(*snd_device_index)) != 0) {
- ALOGE("%s: %s in platform.h at index %d does not match %s, from %s no ACDB ID set!",
- __func__, platform_get_snd_device_name(*snd_device_index),
- *snd_device_index, attr[1], PLATFORM_XML_PATH);
+ 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 at index %d in %s has no acdb_id, no ACDB ID set!",
- __func__, attr[1], *snd_device_index, PLATFORM_XML_PATH);
+ 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(*snd_device_index,
- atoi((char *)attr[3])) != 0)
+ 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;
+ }
-next:
- (*snd_device_index)++;
done:
return;
}
@@ -82,7 +81,7 @@
unsigned int i;
if (strcmp(tag_name, "device") == 0)
- process_device(userdata, attr);
+ process_device(attr);
return;
}
@@ -98,13 +97,12 @@
FILE *file;
int ret = 0;
int bytes_read;
- unsigned int snd_device_index = SND_DEVICE_MIN;
void *buf;
- file = fopen(PLATFORM_XML_PATH, "r");
+ file = fopen(PLATFORM_INFO_XML_PATH, "r");
if (!file) {
ALOGD("%s: Failed to open %s, using defaults.",
- __func__, PLATFORM_XML_PATH);
+ __func__, PLATFORM_INFO_XML_PATH);
ret = -ENODEV;
goto done;
}
@@ -116,7 +114,6 @@
goto err_close_file;
}
- XML_SetUserData(parser, &snd_device_index);
XML_SetElementHandler(parser, start_tag, end_tag);
while (1) {
@@ -137,7 +134,7 @@
if (XML_ParseBuffer(parser, bytes_read,
bytes_read == 0) == XML_STATUS_ERROR) {
ALOGE("%s: XML_ParseBuffer failed, for %s",
- __func__, PLATFORM_XML_PATH);
+ __func__, PLATFORM_INFO_XML_PATH);
ret = -EINVAL;
goto err_free_parser;
}
@@ -146,12 +143,6 @@
break;
}
- if (snd_device_index != SND_DEVICE_MAX) {
- ALOGE("%s: Only %d/%d ACDB ID's set! Fix %s!",
- __func__, snd_device_index, SND_DEVICE_MAX, PLATFORM_XML_PATH);
- ret = -EINVAL;
- }
-
err_free_parser:
XML_ParserFree(parser);
err_close_file:
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;
}