Merge "hal: Read hotword data from sound trigger hal"
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index e5201cc..391a501 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -214,7 +214,7 @@
snprintf(name, MAX_PATH, TZ_TYPE, tzn);
ALOGD("Opening %s\n", name);
read_line_from_file(name, buf, sizeof(buf));
- buf[strlen(sensor_name)] = '\0';
+ buf[strlen(buf)] = '\0';
if (!strcmp(buf, sensor_name)) {
found = 1;
break;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 297d671..ae4de62 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -626,7 +626,7 @@
(usecase->out_snd_device != snd_device || force_routing) &&
usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&
usecase_backend_idx == backend_idx) {
- ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", __func__,
+ ALOGD("%s: Usecase (%s) is active on (%s) - disabling ..", __func__,
use_case_table[usecase->id],
platform_get_snd_device_name(usecase->out_snd_device));
disable_audio_route(adev, usecase);
@@ -870,7 +870,6 @@
usecase->stream.out);
if (usecase->stream.out == adev->primary_output &&
adev->active_input &&
- adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
out_snd_device != usecase->out_snd_device) {
select_devices(adev, adev->active_input->usecase);
}
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index e6503dd..e2a9648 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -38,7 +38,7 @@
#include "sound/msmcal-hwdep.h"
#include <dirent.h>
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
-
+#define MAX_MIXER_XML_PATH 100
#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
#define MIXER_XML_PATH_MTP "/system/etc/mixer_paths_mtp.xml"
#define MIXER_XML_PATH_SBC "/system/etc/mixer_paths_sbc.xml"
@@ -58,6 +58,7 @@
#define MIXER_XML_PATH_WCD9306 "/system/etc/mixer_paths_wcd9306.xml"
#define MIXER_XML_PATH_WCD9330 "/system/etc/mixer_paths_wcd9330.xml"
#define MIXER_XML_PATH_WCD9335 "/system/etc/mixer_paths_wcd9335.xml"
+#define MIXER_XML_PATH_WCD9326 "/system/etc/mixer_paths_wcd9326.xml"
#define MIXER_XML_PATH_SKUN "/system/etc/mixer_paths_qrd_skun.xml"
#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define PLATFORM_INFO_XML_PATH_EXTCODEC "/system/etc/audio_platform_info_extcodec.xml"
@@ -136,7 +137,6 @@
char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
[WCD9XXX_ANC_CAL] = "anc_cal",
[WCD9XXX_MBHC_CAL] = "mbhc_cal",
- [WCD9XXX_MAD_CAL] = "mad_cal",
};
#define AUDIO_PARAMETER_KEY_REC_PLAY_CONC "rec_play_conc_on"
@@ -746,6 +746,8 @@
sizeof("msm8952-tomtom-snd-card")) ||
!strncmp(snd_card_name, "msm8976-tasha-snd-card",
sizeof("msm8976-tasha-snd-card")) ||
+ !strncmp(snd_card_name, "msm8976-tashalite-snd-card",
+ sizeof("msm8976-tashalite-snd-card")) ||
!strncmp(snd_card_name, "msm8976-tasha-skun-snd-card",
sizeof("msm8976-tasha-skun-snd-card")))
{
@@ -851,6 +853,14 @@
msm_be_id_array_len =
sizeof(msm_device_to_be_id_external_codec) / sizeof(msm_device_to_be_id_external_codec[0]);
+ } else if (!strncmp(snd_card_name, "msm8976-tashalite-snd-card",
+ sizeof("msm8976-tashalite-snd-card"))) {
+ strlcpy(mixer_xml_path, MIXER_XML_PATH_WCD9326,
+ MAX_MIXER_XML_PATH);
+ msm_device_to_be_id = msm_device_to_be_id_external_codec;
+ msm_be_id_array_len =
+ sizeof(msm_device_to_be_id_external_codec) / sizeof(msm_device_to_be_id_external_codec[0]);
+
} else if (!strncmp(snd_card_name, "msm8976-tasha-skun-snd-card",
sizeof("msm8976-tasha-skun-snd-card"))) {
strlcpy(mixer_xml_path, MIXER_XML_PATH_SKUN,
@@ -1188,8 +1198,6 @@
struct wcdcal_ioctl_buffer codec_buffer;
struct param_data calib;
- if (!strcmp(cal_name_info[type], "mad_cal"))
- calib.acdb_id = SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID;
calib.get_size = 1;
ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
&calib);
@@ -1286,7 +1294,7 @@
struct platform_data *my_data = NULL;
int retry_num = 0, snd_card_num = 0, key = 0;
const char *snd_card_name;
- char mixer_xml_path[100],ffspEnable[PROPERTY_VALUE_MAX];
+ char mixer_xml_path[MAX_MIXER_XML_PATH],ffspEnable[PROPERTY_VALUE_MAX];
char *cvd_version = NULL;
const char *mixer_ctl_name = "Set HPX ActiveBe";
struct mixer_ctl *ctl = NULL;
@@ -1802,7 +1810,7 @@
{
if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
ALOGE("%s: Invalid snd_device = %d", __func__, snd_device);
- return DEFAULT_OUTPUT_SAMPLING_RATE;
+ return CODEC_BACKEND_DEFAULT_BIT_WIDTH;
}
return backend_bit_width_table[snd_device];
}
@@ -1811,7 +1819,7 @@
{
na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
= codec_support;
- ALOGV("%s: na_props.platform_na_prop_enabled: %d", __func__,
+ ALOGD("%s: na_props.platform_na_prop_enabled: %d", __func__,
na_props.platform_na_prop_enabled);
return 0;
}
@@ -1862,10 +1870,15 @@
value, len);
if (ret >= 0) {
if (na_props.platform_na_prop_enabled) {
- if (!strncmp("true", value, sizeof("true")))
+ if (!strncmp("true", value, sizeof("true"))) {
na_props.ui_na_prop_enabled = true;
- else
+ ALOGD("%s: native audio feature enabled from UI",__func__);
+ }
+ else {
na_props.ui_na_prop_enabled = false;
+ ALOGD("%s: native audio feature disabled from UI",__func__);
+
+ }
str_parms_del(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO);
@@ -1878,14 +1891,15 @@
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) &&
OUTPUT_SAMPLING_RATE_44100 == usecase->stream.out->sample_rate) {
- select_devices(platform->adev, usecase->id);
- ALOGV("%s: triggering dynamic device switch for usecase: "
- "%d, device: %d", __func__, usecase->id,
+ ALOGD("%s: triggering dynamic device switch for usecase(%d: %s)"
+ " stream(%p), device(%d)", __func__, usecase->id,
+ use_case_table[usecase->id], usecase->stream,
usecase->stream.out->devices);
+ select_devices(platform->adev, usecase->id);
}
}
} else {
- ALOGV("%s: native audio not supported: %d", __func__,
+ ALOGD("%s: native audio not supported: %d", __func__,
na_props.platform_na_prop_enabled);
}
}
@@ -3564,6 +3578,8 @@
sizeof("msm8952-tomtom-snd-card")) ||
!strncmp(snd_card_name, "msm8976-tasha-snd-card",
sizeof("msm8976-tasha-snd-card")) ||
+ !strncmp(snd_card_name, "msm8976-tashalite-snd-card",
+ sizeof("msm8976-tashalite-snd-card")) ||
!strncmp(snd_card_name, "msm8976-tasha-skun-snd-card",
sizeof("msm8976-tasha-skun-snd-card")))
{
@@ -3768,6 +3784,16 @@
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
}
+
+ /*
+ * Sample rate greater than 48K is only supported by external codecs on
+ * specific devices e.g. Headphones, reset the sample rate to
+ * default value if not external codec.
+ */
+ if (!is_external_codec)
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+
+
ALOGI("%s Codec selected backend: %d updated bit width: %d and sample rate: %d",
__func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 5b339a7..019a889 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -129,7 +129,6 @@
char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
[WCD9XXX_ANC_CAL] = "anc_cal",
[WCD9XXX_MBHC_CAL] = "mbhc_cal",
- [WCD9XXX_MAD_CAL] = "mad_cal",
};
enum {
@@ -982,8 +981,6 @@
struct wcdcal_ioctl_buffer codec_buffer;
struct param_data calib;
- if (!strcmp(cal_name_info[type], "mad_cal"))
- calib.acdb_id = SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID;
calib.get_size = 1;
ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
&calib);
@@ -1634,7 +1631,7 @@
{
na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
= codec_support;
- ALOGV("%s: na_props.platform_na_prop_enabled: %d", __func__,
+ ALOGD("%s: na_props.platform_na_prop_enabled: %d", __func__,
na_props.platform_na_prop_enabled);
return 0;
}
@@ -1685,10 +1682,15 @@
value, len);
if (ret >= 0) {
if (na_props.platform_na_prop_enabled) {
- if (!strncmp("true", value, sizeof("true")))
+ if (!strncmp("true", value, sizeof("true"))) {
na_props.ui_na_prop_enabled = true;
- else
+ ALOGD("%s: native audio feature enabled from UI",__func__);
+ }
+ else {
na_props.ui_na_prop_enabled = false;
+ ALOGD("%s: native audio feature disabled from UI",__func__);
+
+ }
str_parms_del(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO);
@@ -1703,14 +1705,15 @@
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) &&
OUTPUT_SAMPLING_RATE_44100 == usecase->stream.out->sample_rate) {
- select_devices(platform->adev, usecase->id);
- ALOGV("%s: triggering dynamic device switch for usecase: "
- "%d, device: %d", __func__, usecase->id,
+ ALOGD("%s: triggering dynamic device switch for usecase(%d: %s)"
+ " stream(%p), device(%d)", __func__, usecase->id,
+ use_case_table[usecase->id], usecase->stream,
usecase->stream.out->devices);
+ select_devices(platform->adev, usecase->id);
}
}
} else {
- ALOGV("%s: native audio not supported: %d", __func__,
+ ALOGD("%s: native audio not supported: %d", __func__,
na_props.platform_na_prop_enabled);
}
}
diff --git a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
index 4cfee1b..6154e0c 100644
--- a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
+++ b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
@@ -4070,6 +4070,8 @@
//The total length of the data to be transcoded
srcStart = buffer->pBuffer;
OMX_U8 *data = NULL;
+ ssize_t bytes = 0;
+
PrintFrameHdr(OMX_COMPONENT_GENERATE_ETB,buffer);
memset(&meta_in,0,sizeof(meta_in));
if ( search_input_bufhdr(buffer) == false )
@@ -4104,7 +4106,22 @@
}
memcpy(&data[sizeof(META_IN)],buffer->pBuffer,buffer->nFilledLen);
- write(m_drv_fd, data, buffer->nFilledLen+sizeof(META_IN));
+ bytes = write(m_drv_fd, data, buffer->nFilledLen+sizeof(META_IN));
+ if (bytes <= 0) {
+ frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+ if (errno == ENETRESET)
+ {
+ ALOGE("In SSR, return error to close the session");
+ m_cb.EventHandler(&m_cmp,
+ m_app_data,
+ OMX_EventError,
+ OMX_ErrorHardware,
+ 0, NULL );
+ }
+ return OMX_ErrorNone;
+ }
+
pthread_mutex_lock(&m_state_lock);
get_state(&m_cmp, &state);
pthread_mutex_unlock(&m_state_lock);
diff --git a/mm-audio/aenc-evrc/qdsp6/src/omx_evrc_aenc.cpp b/mm-audio/aenc-evrc/qdsp6/src/omx_evrc_aenc.cpp
index 8200365..af9f785 100644
--- a/mm-audio/aenc-evrc/qdsp6/src/omx_evrc_aenc.cpp
+++ b/mm-audio/aenc-evrc/qdsp6/src/omx_evrc_aenc.cpp
@@ -3974,6 +3974,8 @@
//The total length of the data to be transcoded
srcStart = buffer->pBuffer;
OMX_U8 *data = NULL;
+ ssize_t bytes = 0;
+
PrintFrameHdr(OMX_COMPONENT_GENERATE_ETB,buffer);
memset(&meta_in,0,sizeof(meta_in));
if ( search_input_bufhdr(buffer) == false )
@@ -4003,7 +4005,21 @@
}
memcpy(&data[sizeof(META_IN)],buffer->pBuffer,buffer->nFilledLen);
- write(m_drv_fd, data, buffer->nFilledLen+sizeof(META_IN));
+ bytes = write(m_drv_fd, data, buffer->nFilledLen+sizeof(META_IN));
+ if (bytes <= 0) {
+ frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+ if (errno == ENETRESET)
+ {
+ ALOGE("In SSR, return error to close the session");
+ m_cb.EventHandler(&m_cmp,
+ m_app_data,
+ OMX_EventError,
+ OMX_ErrorHardware,
+ 0, NULL );
+ }
+ return OMX_ErrorNone;
+ }
pthread_mutex_lock(&m_state_lock);
get_state(&m_cmp, &state);
@@ -4045,11 +4061,21 @@
buffer->nAllocLen,buffer->pBuffer,
nReadbytes,nNumOutputBuf);
if (nReadbytes <= 0) {
- buffer->nFilledLen = 0;
+ buffer->nFilledLen = 0;
buffer->nOffset = 0;
- buffer->nTimeStamp = nTimestamp;
- frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
- return OMX_ErrorNone;
+ buffer->nTimeStamp = nTimestamp;
+ frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+ if (errno == ENETRESET)
+ {
+ ALOGE("In SSR, return error to close the session");
+ m_cb.EventHandler(&m_cmp,
+ m_app_data,
+ OMX_EventError,
+ OMX_ErrorHardware,
+ 0, NULL );
+ }
+ return OMX_ErrorNone;
} else
DEBUG_PRINT("Read bytes %d\n",nReadbytes);
// Buffer from Driver will have
diff --git a/mm-audio/aenc-qcelp13/qdsp6/src/omx_qcelp13_aenc.cpp b/mm-audio/aenc-qcelp13/qdsp6/src/omx_qcelp13_aenc.cpp
index 399b8cf..d25eb7f 100644
--- a/mm-audio/aenc-qcelp13/qdsp6/src/omx_qcelp13_aenc.cpp
+++ b/mm-audio/aenc-qcelp13/qdsp6/src/omx_qcelp13_aenc.cpp
@@ -3972,6 +3972,8 @@
//The total length of the data to be transcoded
srcStart = buffer->pBuffer;
OMX_U8 *data = NULL;
+ ssize_t bytes = 0;
+
PrintFrameHdr(OMX_COMPONENT_GENERATE_ETB,buffer);
memset(&meta_in,0,sizeof(meta_in));
if ( search_input_bufhdr(buffer) == false )
@@ -4001,7 +4003,21 @@
}
memcpy(&data[sizeof(META_IN)],buffer->pBuffer,buffer->nFilledLen);
- write(m_drv_fd, data, buffer->nFilledLen+sizeof(META_IN));
+ bytes = write(m_drv_fd, data, buffer->nFilledLen+sizeof(META_IN));
+ if (bytes <= 0) {
+ frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+ if (errno == ENETRESET)
+ {
+ ALOGE("In SSR, return error to close the session");
+ m_cb.EventHandler(&m_cmp,
+ m_app_data,
+ OMX_EventError,
+ OMX_ErrorHardware,
+ 0, NULL );
+ }
+ return OMX_ErrorNone;
+ }
pthread_mutex_lock(&m_state_lock);
get_state(&m_cmp, &state);
@@ -4043,11 +4059,21 @@
buffer->nAllocLen,buffer->pBuffer,
nReadbytes,nNumOutputBuf);
if (nReadbytes <= 0) {
- buffer->nFilledLen = 0;
+ buffer->nFilledLen = 0;
buffer->nOffset = 0;
- buffer->nTimeStamp = nTimestamp;
- frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
- return OMX_ErrorNone;
+ buffer->nTimeStamp = nTimestamp;
+ frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+ if (errno == ENETRESET)
+ {
+ ALOGE("In SSR, return error to close the session");
+ m_cb.EventHandler(&m_cmp,
+ m_app_data,
+ OMX_EventError,
+ OMX_ErrorHardware,
+ 0, NULL );
+ }
+ return OMX_ErrorNone;
} else
DEBUG_PRINT("Read bytes %d\n",nReadbytes);
diff --git a/post_proc/EffectsHwAcc.cpp b/post_proc/EffectsHwAcc.cpp
index 0e4c55a..e11cfc7 100644
--- a/post_proc/EffectsHwAcc.cpp
+++ b/post_proc/EffectsHwAcc.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-15, 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
@@ -72,7 +72,7 @@
size_t reqOutputFrameCount = pBuffer->frameCount;
int ret = 0;
- if (mTrackBufferProvider != NULL) {
+ if (mTrackInputBufferProvider != NULL) {
while (1) {
reqInputFrameCount = ((reqOutputFrameCount *
mEffectsConfig.inputCfg.samplingRate)/
@@ -89,7 +89,7 @@
popcount(mEffectsConfig.inputCfg.channels);
while (frameCount) {
pBuffer->frameCount = frameCount;
- ret = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+ ret = mTrackInputBufferProvider->getNextBuffer(pBuffer, pts);
if (ret == OK) {
int bytesInBuffer = pBuffer->frameCount *
FRAME_SIZE(mEffectsConfig.inputCfg.format) *
@@ -98,7 +98,7 @@
frameCount -= pBuffer->frameCount;
mInputBufferFrameCountOffset += pBuffer->frameCount;
offset += bytesInBuffer;
- mTrackBufferProvider->releaseBuffer(pBuffer);
+ mTrackInputBufferProvider->releaseBuffer(pBuffer);
} else
break;
}
@@ -133,7 +133,7 @@
AudioBufferProvider::Buffer *pBuffer)
{
ALOGV("EffBufferProvider::releaseBuffer()");
- if (this->mTrackBufferProvider != NULL) {
+ if (this->mTrackInputBufferProvider != NULL) {
pBuffer->frameCount = 0;
pBuffer->raw = NULL;
} else {
@@ -189,7 +189,8 @@
mEnabled = false;
}
-status_t EffectsHwAcc::prepareEffects(AudioBufferProvider **bufferProvider,
+status_t EffectsHwAcc::prepareEffects(AudioBufferProvider **inputBufferProvider,
+ AudioBufferProvider **bufferProvider,
int sessionId,
audio_channel_mask_t channelMask,
int frameCount)
@@ -316,10 +317,11 @@
goto noEffectsForActiveTrack;
}
// initialization successful:
- // - keep track of the real buffer provider in case it was set before
+ // - keep backup of track's buffer provider
pHwAccbp->mTrackBufferProvider = *bufferProvider;
- // - we'll use the hw acc effect integrated inside this
- // track's buffer provider, and we'll use it as the track's buffer provider
+ pHwAccbp->mTrackInputBufferProvider = *inputBufferProvider;
+ // - we'll use the hw acc effect integrated inside this track's buffer provider,
+ // and we'll use it as the track's buffer provider
mBufferProvider = pHwAccbp;
*bufferProvider = pHwAccbp;
@@ -332,14 +334,14 @@
return NO_INIT;
}
-void EffectsHwAcc::setBufferProvider(AudioBufferProvider **bufferProvider,
+void EffectsHwAcc::setBufferProvider(AudioBufferProvider **trackInputBufferProvider,
AudioBufferProvider **trackBufferProvider)
{
ALOGV("setBufferProvider");
if (mBufferProvider &&
- (mBufferProvider->mTrackBufferProvider != *bufferProvider)) {
+ (mBufferProvider->mTrackInputBufferProvider != *trackInputBufferProvider)) {
*trackBufferProvider = mBufferProvider;
- mBufferProvider->mTrackBufferProvider = *bufferProvider;
+ mBufferProvider->mTrackInputBufferProvider = *trackInputBufferProvider;
}
}
diff --git a/post_proc/EffectsHwAcc.h b/post_proc/EffectsHwAcc.h
index 6420a9b..0452f57 100644
--- a/post_proc/EffectsHwAcc.h
+++ b/post_proc/EffectsHwAcc.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-15, 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
@@ -43,10 +43,11 @@
virtual void setSampleRate(uint32_t inpSR, uint32_t outSR);
virtual void unprepareEffects(AudioBufferProvider **trackBufferProvider);
- virtual status_t prepareEffects(AudioBufferProvider **trackBufferProvider,
+ virtual status_t prepareEffects(AudioBufferProvider **trackInputBufferProvider,
+ AudioBufferProvider **trackBufferProvider,
int sessionId, audio_channel_mask_t channelMask,
int frameCount);
- virtual void setBufferProvider(AudioBufferProvider **bufferProvider,
+ virtual void setBufferProvider(AudioBufferProvider **trackInputbufferProvider,
AudioBufferProvider **trackBufferProvider);
#ifdef HW_ACC_HPX
virtual void updateHPXState(uint32_t state);
@@ -62,6 +63,7 @@
virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
virtual void releaseBuffer(Buffer* buffer);
+ AudioBufferProvider* mTrackInputBufferProvider;
AudioBufferProvider* mTrackBufferProvider;
effect_handle_t mEffectsHandle;
effect_config_t mEffectsConfig;
diff --git a/post_proc/reverb.c b/post_proc/reverb.c
index b256e53..450ce81 100644
--- a/post_proc/reverb.c
+++ b/post_proc/reverb.c
@@ -281,17 +281,19 @@
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->enabled_by_client) {
+ 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,
+ if (context->ctl)
+ offload_reverb_send_params(context->ctl, &context->offload_reverb,
OFFLOAD_SEND_REVERB_ENABLE_FLAG |
OFFLOAD_SEND_REVERB_PRESET);
- if (context->hw_acc_fd > 0)
- hw_acc_reverb_send_params(context->hw_acc_fd, &context->offload_reverb,
+ if (context->hw_acc_fd > 0)
+ hw_acc_reverb_send_params(context->hw_acc_fd, &context->offload_reverb,
OFFLOAD_SEND_REVERB_ENABLE_FLAG |
OFFLOAD_SEND_REVERB_PRESET);
+ }
}
void reverb_set_all_properties(reverb_context_t *context,
@@ -600,6 +602,7 @@
set_config(context, &context->config);
reverb_ctxt->hw_acc_fd = -1;
+ reverb_ctxt->enabled_by_client = false;
memset(&(reverb_ctxt->reverb_settings), 0, sizeof(reverb_settings_t));
memset(&(reverb_ctxt->offload_reverb), 0, sizeof(struct reverb_params));
@@ -615,6 +618,16 @@
reverb_context_t *reverb_ctxt = (reverb_context_t *)context;
ALOGV("%s: ctxt %p", __func__, reverb_ctxt);
+ reverb_ctxt->enabled_by_client = true;
+
+ /* REVERB_PRESET_NONE is equivalent to disabled state,
+ * But support for this state is not provided in DSP.
+ * Hence, do not set enable flag, if in peset mode with preset "NONE".
+ * Effect would be enabled when valid preset is set.
+ */
+ if ((reverb_ctxt->preset == true) &&
+ (reverb_ctxt->next_preset == REVERB_PRESET_NONE))
+ return 0;
if (!offload_reverb_get_enable_flag(&(reverb_ctxt->offload_reverb)))
offload_reverb_set_enable_flag(&(reverb_ctxt->offload_reverb), true);
@@ -626,6 +639,7 @@
reverb_context_t *reverb_ctxt = (reverb_context_t *)context;
ALOGV("%s: ctxt %p", __func__, reverb_ctxt);
+ reverb_ctxt->enabled_by_client = false;
if (offload_reverb_get_enable_flag(&(reverb_ctxt->offload_reverb))) {
offload_reverb_set_enable_flag(&(reverb_ctxt->offload_reverb), false);
if (reverb_ctxt->ctl)
diff --git a/post_proc/reverb.h b/post_proc/reverb.h
index 991151e..1a5ca0d 100644
--- a/post_proc/reverb.h
+++ b/post_proc/reverb.h
@@ -48,6 +48,7 @@
// Offload vars
struct mixer_ctl *ctl;
int hw_acc_fd;
+ bool enabled_by_client;
bool auxiliary;
bool preset;
uint16_t cur_preset;
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 2748568..3874f0b 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -56,6 +56,15 @@
ALOGV("%s: ctxt %p, strength: %d", __func__, context, strength);
context->strength = strength;
+ /*
+ * Zero strength is not equivalent to disable state as down mix
+ * is still happening for multichannel inputs.
+ * For better user experience, explicitly disable virtualizer module
+ * when strength is 0.
+ */
+ offload_virtualizer_set_enable_flag(&(context->offload_virt),
+ ((strength > 0) && !(context->temp_disabled)) ?
+ true : false);
offload_virtualizer_set_strength(&(context->offload_virt), strength);
if (context->ctl)
offload_virtualizer_send_params(context->ctl, &context->offload_virt,