Merge remote-tracking branch 'goog/stage-aosp-master' into HEAD
am: a350867cd5
Change-Id: I7c9e7311269cd9acff74475e1140b6583c266660
diff --git a/Android.mk b/Android.mk
index 65bf578..d6f03fb 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,6 +1,6 @@
# TODO: Find a better way to separate build configs for ADP vs non-ADP devices
ifneq ($(TARGET_BOARD_AUTO),true)
- ifneq ($(filter msm8960 msm8226 msm8x26 msm8x84 msm8084 msm8992 msm8994 msm8996 msm8909 msm8952 msm8998,$(TARGET_BOARD_PLATFORM)),)
+ ifneq ($(filter msm8960 msm8226 msm8x26 msm8x84 msm8084 msm8992 msm8994 msm8996 msm8909 msm8952 msm8998 sdm845,$(TARGET_BOARD_PLATFORM)),)
MY_LOCAL_PATH := $(call my-dir)
diff --git a/hal/Android.mk b/hal/Android.mk
index 923879c..58a79fb 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -10,7 +10,7 @@
ifneq ($(filter msm8960,$(TARGET_BOARD_PLATFORM)),)
LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="2"
endif
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msm8998,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msm8998 sdm845,$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM = msm8974
ifneq ($(filter msm8974,$(TARGET_BOARD_PLATFORM)),)
@@ -47,6 +47,12 @@
LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
MULTIPLE_HW_VARIANTS_ENABLED := true
endif
+ifneq ($(filter sdm845,$(TARGET_BOARD_PLATFORM)),)
+ LOCAL_CFLAGS := -DPLATFORM_SDM845
+ LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+ LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
+ MULTIPLE_HW_VARIANTS_ENABLED := true
+endif
endif
ifneq ($(filter msm8916 msm8909 msm8952,$(TARGET_BOARD_PLATFORM)),)
@@ -149,7 +155,7 @@
LOCAL_SRC_FILES += audio_extn/dsm_feedback.c
endif
-ifneq ($(filter msm8992 msm8994 msm8996 msm8998,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8998 sdm845,$(TARGET_BOARD_PLATFORM)),)
# push codec/mad calibration to HW dep node
# applicable to msm8992/8994 or newer platforms
LOCAL_CFLAGS += -DHWDEP_CAL_ENABLED
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index c7d6768..2ca76fa 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -81,7 +81,7 @@
#define audio_extn_usb_is_capture_supported() (false)
#define audio_extn_usb_get_max_channels(dir) (0)
#define audio_extn_usb_get_max_bit_width(dir) (0)
-#define audio_extn_usb_sup_sample_rates(t, s, l) (0)
+#define audio_extn_usb_sup_sample_rates(t, s, l) ((t), (s), (l), 0) /* fix unused warn */
#define audio_extn_usb_alive(adev) (false)
#else
void audio_extn_usb_init(void *adev);
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 07a9711..ad1530a 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -438,10 +438,17 @@
ret = str_parms_get_str(parms, AUDIO_PARAMETER_HFP_ENABLE, value,
sizeof(value));
if (ret >= 0) {
- if (!strncmp(value,"true",sizeof(value)))
- ret = start_hfp(adev,parms);
- else
- stop_hfp(adev);
+ if (!strncmp(value,"true",sizeof(value))) {
+ if (!hfpmod.is_hfp_running)
+ start_hfp(adev,parms);
+ else
+ ALOGW("%s: HFP is already active.", __func__);
+ } else {
+ if (hfpmod.is_hfp_running)
+ stop_hfp(adev);
+ else
+ ALOGW("%s: ignore STOP, HFC not active", __func__);
+ }
}
memset(value, 0, sizeof(value));
ret = str_parms_get_str(parms,AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE, value,
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 8afb0dc..22920b7 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -51,7 +51,7 @@
#define MAX_RESISTANCE_SPKR_Q24 (40 * (1 << 24))
/*Path where the calibration file will be stored*/
-#define CALIB_FILE "/data/misc/audio/audio.cal"
+#define CALIB_FILE "/data/vendor/audio/audio.cal"
/*Time between retries for calibartion or intial wait time
after boot up*/
@@ -304,6 +304,7 @@
int32_t pcm_dev_rx_id = -1, pcm_dev_tx_id = -1;
struct timespec ts;
int retry_duration;
+ int app_type = 0;
if (!adev) {
ALOGE("%s: Invalid params", __func__);
@@ -472,7 +473,20 @@
handle.pcm_tx = NULL;
/* Clear TX calibration to handset mic */
- platform_send_audio_calibration(adev->platform, SND_DEVICE_IN_HANDSET_MIC);
+ if (platform_supports_app_type_cfg()) {
+ ALOGD("%s: Platform supports APP type configuration, using V2\n", __func__);
+ if (uc_info_tx != NULL) {
+ ALOGD("%s: UC Info TX is not NULL, updating and sending calibration\n", __func__);
+ uc_info_tx->in_snd_device = SND_DEVICE_IN_HANDSET_MIC;
+ uc_info_tx->out_snd_device = SND_DEVICE_NONE;
+ platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE, &app_type);
+ platform_send_audio_calibration_v2(adev->platform, uc_info_tx,
+ app_type, 8000);
+ }
+ } else {
+ ALOGW("%s: Platform does NOT support APP type configuration, using V1\n", __func__);
+ platform_send_audio_calibration(adev->platform, SND_DEVICE_IN_HANDSET_MIC);
+ }
if (!status.status) {
protCfg.mode = MSM_SPKR_PROT_CALIBRATED;
protCfg.r0[SP_V2_SPKR_1] = status.r0[SP_V2_SPKR_1];
@@ -804,6 +818,7 @@
struct audio_usecase *uc_info_tx;
struct audio_device *adev = handle.adev_handle;
int32_t pcm_dev_tx_id = -1, ret = 0;
+ int app_type = 0;
ALOGV("%s: Entry", __func__);
if (!adev) {
@@ -855,7 +870,20 @@
exit:
/* Clear VI feedback cal and replace with handset MIC */
- platform_send_audio_calibration(adev->platform, SND_DEVICE_IN_HANDSET_MIC);
+ if (platform_supports_app_type_cfg()) {
+ ALOGD("%s: Platform supports APP type configuration, using V2\n", __func__);
+ if (uc_info_tx != NULL) {
+ ALOGD("%s: UC Info TX is not NULL, updating and sending calibration\n", __func__);
+ uc_info_tx->in_snd_device = SND_DEVICE_IN_HANDSET_MIC;
+ uc_info_tx->out_snd_device = SND_DEVICE_NONE;
+ platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE, &app_type);
+ platform_send_audio_calibration_v2(adev->platform, uc_info_tx,
+ app_type, 8000);
+ }
+ } else {
+ ALOGW("%s: Platform does not support APP type configuration, using V1\n", __func__);
+ platform_send_audio_calibration(adev->platform, SND_DEVICE_IN_HANDSET_MIC);
+ }
if (ret) {
if (handle.pcm_tx)
pcm_close(handle.pcm_tx);
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 7267931..f8c17d0 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -999,9 +999,9 @@
return ret;
}
-static ssize_t read_usb_sup_sample_rates(bool is_playback __unused,
- uint32_t *supported_sample_rates __unused,
- uint32_t max_rates __unused)
+static ssize_t read_usb_sup_sample_rates(bool is_playback,
+ uint32_t *supported_sample_rates,
+ uint32_t max_rates)
{
ssize_t count = audio_extn_usb_sup_sample_rates(is_playback,
supported_sample_rates,
@@ -3680,18 +3680,17 @@
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
- int i, ret;
+ int i, ret = 0;
bool is_hdmi = devices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
bool is_usb_dev = audio_is_usb_out_device(devices) &&
(devices != AUDIO_DEVICE_OUT_USB_ACCESSORY);
- bool direct_dev = is_hdmi || is_usb_dev;
if (is_usb_dev && !is_usb_ready(adev, true /* is_playback */)) {
return -ENOSYS;
}
- ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
- __func__, config->sample_rate, config->channel_mask, devices, flags);
+ ALOGV("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
+ __func__, config->format, config->sample_rate, config->channel_mask, devices, flags);
*stream_out = NULL;
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
@@ -3701,16 +3700,14 @@
out->flags = flags;
out->devices = devices;
out->dev = adev;
- out->format = config->format;
- out->sample_rate = config->sample_rate;
- out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
out->handle = handle;
/* Init use case and pcm_config */
- if (audio_is_linear_pcm(out->format) &&
- (out->flags == AUDIO_OUTPUT_FLAG_NONE ||
- out->flags == AUDIO_OUTPUT_FLAG_DIRECT) && direct_dev) {
+ if ((is_hdmi || is_usb_dev) &&
+ (audio_is_linear_pcm(config->format) || config->format == AUDIO_FORMAT_DEFAULT) &&
+ (flags == AUDIO_OUTPUT_FLAG_NONE ||
+ (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)) {
pthread_mutex_lock(&adev->lock);
if (is_hdmi) {
ret = read_hdmi_channel_masks(out);
@@ -3732,25 +3729,38 @@
&out->supported_sample_rates[0],
MAX_SUPPORTED_SAMPLE_RATES);
ALOGV("plugged dev USB ret %d", ret);
- } else {
- ret = -1;
}
pthread_mutex_unlock(&adev->lock);
if (ret != 0)
goto error_open;
- out->channel_mask = config->channel_mask;
+
out->sample_rate = config->sample_rate;
+ out->channel_mask = config->channel_mask;
out->format = config->format;
- out->usecase = USECASE_AUDIO_PLAYBACK_HIFI;
- // does this change?
- out->config = is_hdmi ? pcm_config_hdmi_multi : pcm_config_hifi;
- out->config.rate = config->sample_rate;
+ if (is_hdmi) {
+ out->usecase = USECASE_AUDIO_PLAYBACK_HIFI;
+ out->config = pcm_config_hdmi_multi;
+ } else if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
+ out->usecase = USECASE_AUDIO_PLAYBACK_MMAP;
+ out->config = pcm_config_mmap_playback;
+ out->stream.start = out_start;
+ out->stream.stop = out_stop;
+ out->stream.create_mmap_buffer = out_create_mmap_buffer;
+ out->stream.get_mmap_position = out_get_mmap_position;
+ } else {
+ out->usecase = USECASE_AUDIO_PLAYBACK_HIFI;
+ out->config = pcm_config_hifi;
+ }
+
+ out->config.rate = out->sample_rate;
out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
- out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels *
- audio_bytes_per_sample(config->format));
+ if (is_hdmi) {
+ out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels *
+ audio_bytes_per_sample(out->format));
+ }
out->config.format = pcm_format_from_audio_format(out->format);
- } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
pthread_mutex_lock(&adev->lock);
bool offline = (adev->card_status == CARD_STATUS_OFFLINE);
pthread_mutex_unlock(&adev->lock);
@@ -3773,17 +3783,20 @@
ret = -EINVAL;
goto error_open;
}
+ out->sample_rate = config->offload_info.sample_rate;
+ if (config->offload_info.channel_mask != AUDIO_CHANNEL_NONE)
+ out->channel_mask = config->offload_info.channel_mask;
+ else if (config->channel_mask != AUDIO_CHANNEL_NONE)
+ out->channel_mask = config->channel_mask;
+ else
+ out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
+ out->format = config->offload_info.format;
out->compr_config.codec = (struct snd_codec *)
calloc(1, sizeof(struct snd_codec));
out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
- if (config->offload_info.channel_mask)
- out->channel_mask = config->offload_info.channel_mask;
- else if (config->channel_mask)
- out->channel_mask = config->channel_mask;
- out->format = config->offload_info.format;
- out->sample_rate = config->offload_info.sample_rate;
out->stream.set_callback = out_set_callback;
out->stream.pause = out_pause;
@@ -3795,11 +3808,11 @@
get_snd_codec_id(config->offload_info.format);
out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
- out->compr_config.codec->sample_rate = config->offload_info.sample_rate;
+ out->compr_config.codec->sample_rate = out->sample_rate;
out->compr_config.codec->bit_rate =
config->offload_info.bit_rate;
out->compr_config.codec->ch_in =
- audio_channel_count_from_out_mask(config->channel_mask);
+ audio_channel_count_from_out_mask(out->channel_mask);
out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
@@ -3812,46 +3825,136 @@
config->offload_info.bit_rate);
} else if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
switch (config->sample_rate) {
+ case 0:
+ out->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ break;
case 8000:
case 16000:
case 48000:
out->sample_rate = config->sample_rate;
break;
default:
- out->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ ALOGE("%s: Unsupported sampling rate %d for Telephony TX", __func__,
+ config->sample_rate);
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ ret = -EINVAL;
+ break;
}
- out->format = AUDIO_FORMAT_PCM_16_BIT;
+ //FIXME: add support for MONO stream configuration when audioflinger mixer supports it
+ switch (config->channel_mask) {
+ case AUDIO_CHANNEL_NONE:
+ out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ break;
+ case AUDIO_CHANNEL_OUT_STEREO:
+ out->channel_mask = config->channel_mask;
+ break;
+ default:
+ ALOGE("%s: Unsupported channel mask %#x for Telephony TX", __func__,
+ config->channel_mask);
+ config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ ret = -EINVAL;
+ break;
+ }
+ switch (config->format) {
+ case AUDIO_FORMAT_DEFAULT:
+ out->format = AUDIO_FORMAT_PCM_16_BIT;
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ out->format = config->format;
+ break;
+ default:
+ ALOGE("%s: Unsupported format %#x for Telephony TX", __func__,
+ config->format);
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ break;
+ }
+ if (ret != 0)
+ goto error_open;
+
out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
out->config = pcm_config_afe_proxy_playback;
+ out->config.rate = out->sample_rate;
+ out->config.channels =
+ audio_channel_count_from_out_mask(out->channel_mask);
+ out->config.format = pcm_format_from_audio_format(out->format);
adev->voice_tx_output = out;
- } else if (out->flags == AUDIO_OUTPUT_FLAG_VOIP_RX) {
+ } else if (flags == AUDIO_OUTPUT_FLAG_VOIP_RX) {
+ switch (config->sample_rate) {
+ case 0:
+ out->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ break;
+ case 8000:
+ case 16000:
+ case 32000:
+ case 48000:
+ out->sample_rate = config->sample_rate;
+ break;
+ default:
+ ALOGE("%s: Unsupported sampling rate %d for Voip RX", __func__,
+ config->sample_rate);
+ config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ ret = -EINVAL;
+ break;
+ }
//FIXME: add support for MONO stream configuration when audioflinger mixer supports it
+ switch (config->channel_mask) {
+ case AUDIO_CHANNEL_NONE:
+ out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ break;
+ case AUDIO_CHANNEL_OUT_STEREO:
+ out->channel_mask = config->channel_mask;
+ break;
+ default:
+ ALOGE("%s: Unsupported channel mask %#x for Voip RX", __func__,
+ config->channel_mask);
+ config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ ret = -EINVAL;
+ break;
+ }
+ switch (config->format) {
+ case AUDIO_FORMAT_DEFAULT:
+ out->format = AUDIO_FORMAT_PCM_16_BIT;
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ out->format = config->format;
+ break;
+ default:
+ ALOGE("%s: Unsupported format %#x for Voip RX", __func__,
+ config->format);
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ break;
+ }
+ if (ret != 0)
+ goto error_open;
+
uint32_t buffer_size, frame_size;
out->usecase = USECASE_AUDIO_PLAYBACK_VOIP;
out->config = pcm_config_voip;
- out->config.format = pcm_format_from_audio_format(config->format);
- out->config.rate = config->sample_rate;
+ out->config.rate = out->sample_rate;
+ out->config.format = pcm_format_from_audio_format(out->format);
buffer_size = get_stream_buffer_size(VOIP_PLAYBACK_PERIOD_DURATION_MSEC,
- config->sample_rate,
- config->format,
+ out->sample_rate,
+ out->format,
out->config.channels,
false /*is_low_latency*/);
- frame_size = audio_bytes_per_sample(config->format) * out->config.channels;
+ frame_size = audio_bytes_per_sample(out->format) * out->config.channels;
out->config.period_size = buffer_size / frame_size;
out->config.period_count = VOIP_PLAYBACK_PERIOD_COUNT;
out->af_period_multiplier = 1;
} else {
- if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
+ if (flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
out->config = pcm_config_deep_buffer;
- } else if (out->flags & AUDIO_OUTPUT_FLAG_TTS) {
+ } else if (flags & AUDIO_OUTPUT_FLAG_TTS) {
out->usecase = USECASE_AUDIO_PLAYBACK_TTS;
out->config = pcm_config_deep_buffer;
- } else if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
+ } else if (flags & AUDIO_OUTPUT_FLAG_RAW) {
out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL, out->flags);
out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency;
- } else if (out->flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
+ } else if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
out->usecase = USECASE_AUDIO_PLAYBACK_MMAP;
out->config = pcm_config_mmap_playback;
out->stream.start = out_start;
@@ -3862,15 +3965,38 @@
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
out->config = pcm_config_low_latency;
}
- if (config->format != audio_format_from_pcm_format(out->config.format)) {
- out->config.format = pcm_format_from_audio_format(config->format);
+
+ if (config->sample_rate == 0) {
+ out->sample_rate = out->config.rate;
+ } else {
+ out->sample_rate = config->sample_rate;
}
- out->sample_rate = out->config.rate;
+ if (config->channel_mask == AUDIO_CHANNEL_NONE) {
+ out->channel_mask = audio_channel_out_mask_from_count(out->config.channels);
+ } else {
+ out->channel_mask = config->channel_mask;
+ }
+ if (config->format == AUDIO_FORMAT_DEFAULT)
+ out->format = audio_format_from_pcm_format(out->config.format);
+ else if (!audio_is_linear_pcm(config->format)) {
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ goto error_open;
+ } else {
+ out->format = config->format;
+ }
+
+ out->config.rate = out->sample_rate;
+ out->config.channels =
+ audio_channel_count_from_out_mask(out->channel_mask);
+ if (out->format != audio_format_from_pcm_format(out->config.format)) {
+ out->config.format = pcm_format_from_audio_format(out->format);
+ }
}
if ((config->sample_rate != 0 && config->sample_rate != out->sample_rate) ||
(config->format != AUDIO_FORMAT_DEFAULT && config->format != out->format) ||
- (config->channel_mask != 0 && config->channel_mask != out->channel_mask)) {
+ (config->channel_mask != AUDIO_CHANNEL_NONE && config->channel_mask != out->channel_mask)) {
ALOGI("%s: Unsupported output config. sample_rate:%u format:%#x channel_mask:%#x",
__func__, config->sample_rate, config->format, config->channel_mask);
config->sample_rate = out->sample_rate;
@@ -3999,6 +4125,7 @@
out->error_log = NULL;
pthread_cond_destroy(&out->cond);
+ pthread_mutex_destroy(&out->pre_lock);
pthread_mutex_destroy(&out->lock);
free(stream);
ALOGV("%s: exit", __func__);
@@ -4544,6 +4671,9 @@
error_log_destroy(in->error_log);
in->error_log = NULL;
+ pthread_mutex_destroy(&in->pre_lock);
+ pthread_mutex_destroy(&in->lock);
+
free(stream);
return;
@@ -4711,6 +4841,7 @@
}
if (adev->adm_deinit)
adev->adm_deinit(adev->adm_data);
+ pthread_mutex_destroy(&adev->lock);
free(device);
}
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index c113eca..5f056a8 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -34,10 +34,10 @@
#include "audio_extn/tfa_98xx.h"
#include <dirent.h>
#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_MSM8909_PM8916 "/system/etc/mixer_paths_msm8909_pm8916.xml"
-#define MIXER_XML_PATH_L9300 "/system/etc/mixer_paths_l9300.xml"
+#define MIXER_XML_PATH "mixer_paths.xml"
+#define MIXER_XML_PATH_MTP "mixer_paths_mtp.xml"
+#define MIXER_XML_PATH_MSM8909_PM8916 "mixer_paths_msm8909_pm8916.xml"
+#define MIXER_XML_PATH_L9300 "mixer_paths_l9300.xml"
#define LIB_ACDB_LOADER "libacdbloader.so"
#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
@@ -485,6 +485,7 @@
{TO_NAME_INDEX(USECASE_VOICEMMODE1_CALL)},
{TO_NAME_INDEX(USECASE_VOICEMMODE2_CALL)},
{TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
+ {TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO_WB)},
{TO_NAME_INDEX(USECASE_AUDIO_SPKR_CALIB_TX)},
};
@@ -836,6 +837,28 @@
return result;
}
+// Treblized config files will be located in /odm/etc or /vendor/etc.
+static const char *kConfigLocationList[] =
+ {"/odm/etc", "/vendor/etc", "/system/etc"};
+static const int kConfigLocationListSize =
+ (sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0]));
+
+bool resolve_config_file(char file_name[MIXER_PATH_MAX_LENGTH]) {
+ char full_config_path[MIXER_PATH_MAX_LENGTH];
+ for (int i = 0; i < kConfigLocationListSize; i++) {
+ snprintf(full_config_path,
+ MIXER_PATH_MAX_LENGTH,
+ "%s/%s",
+ kConfigLocationList[i],
+ file_name);
+ if (F_OK == access(full_config_path, 0)) {
+ strcpy(file_name, full_config_path);
+ return true;
+ }
+ }
+ return false;
+}
+
void *platform_init(struct audio_device *adev)
{
char platform[PROPERTY_VALUE_MAX] = {0};
@@ -845,6 +868,7 @@
int retry_num = 0, snd_card_num = 0, key = 0;
const char *snd_card_name;
char mixer_xml_path[MAX_MIXER_XML_PATH] = {0};
+ char platform_info_path[MAX_MIXER_XML_PATH] = {0};
char ffspEnable[PROPERTY_VALUE_MAX] = {0};
char *cvd_version = NULL;
int idx;
@@ -897,6 +921,11 @@
ALOGE("%s: Failed to init hardware info", __func__);
} else {
query_platform(snd_card_name, mixer_xml_path);
+ if (!resolve_config_file(mixer_xml_path)) {
+ memset(mixer_xml_path, 0, sizeof(mixer_xml_path));
+ strlcpy(mixer_xml_path, MIXER_XML_PATH, MAX_MIXER_XML_PATH);
+ resolve_config_file(mixer_xml_path);
+ }
ALOGD("%s: mixer path file is %s", __func__,
mixer_xml_path);
adev->audio_route = audio_route_init(snd_card_num,
@@ -1020,7 +1049,9 @@
set_platform_defaults();
/* Initialize ACDB and PCM ID's */
- platform_info_init(PLATFORM_INFO_XML_PATH, my_data);
+ strlcpy(platform_info_path, PLATFORM_INFO_XML_PATH, MAX_MIXER_XML_PATH);
+ resolve_config_file(platform_info_path);
+ platform_info_init(platform_info_path, my_data);
/* Read one time ssr property */
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index c61331a..de81a71 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -277,7 +277,7 @@
stop_record_t stop_record;
};
-#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
-#define PLATFORM_INFO_XML_BASE_STRING "/system/etc/audio_platform_info"
+#define PLATFORM_INFO_XML_PATH "audio_platform_info.xml"
+#define PLATFORM_INFO_XML_BASE_STRING "audio_platform_info"
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 6560d14..4addc80 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -30,7 +30,7 @@
#include "platform.h"
#include "audio_extn.h"
#include <linux/msm_audio.h>
-#if defined (PLATFORM_MSM8996) || (PLATFORM_MSM8998)
+#if defined (PLATFORM_MSM8996) || (PLATFORM_MSM8998) || (PLATFORM_SDM845)
#include <sound/devdep_params.h>
#endif
@@ -120,7 +120,7 @@
bool speaker_lr_swap;
void *acdb_handle;
-#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996) || (PLATFORM_MSM8998)
+#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996) || (PLATFORM_MSM8998) || (PLATFORM_SDM845)
acdb_init_v2_cvd_t acdb_init;
#elif defined (PLATFORM_MSM8084)
acdb_init_v2_t acdb_init;
@@ -720,7 +720,7 @@
inline bool platform_supports_app_type_cfg()
{
-#ifdef PLATFORM_MSM8998
+#if defined (PLATFORM_MSM8998) || (PLATFORM_SDM845)
return true;
#else
return false;
@@ -1146,7 +1146,7 @@
return 0;
}
-#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996) || (PLATFORM_MSM8998)
+#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996) || (PLATFORM_MSM8998) || (PLATFORM_SDM845)
char *cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
if (!cvd_version)
ALOGE("failed to allocate cvd_version");
@@ -1472,7 +1472,7 @@
ALOGV("%s: Could not find the symbol acdb_loader_send_gain_dep_cal from %s",
__func__, LIB_ACDB_LOADER);
-#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996) || (PLATFORM_MSM8998)
+#if defined (PLATFORM_MSM8994) || (PLATFORM_MSM8996) || (PLATFORM_MSM8998) || (PLATFORM_SDM845)
acdb_init_v2_cvd_t acdb_init_local;
acdb_init_local = (acdb_init_v2_cvd_t)dlsym(my_data->acdb_handle,
"acdb_loader_init_v2");
@@ -4112,7 +4112,7 @@
int platform_get_mmap_data_fd(void *platform __unused, int fe_dev __unused, int dir __unused,
int *fd __unused, uint32_t *size __unused)
{
-#if defined (PLATFORM_MSM8996) || (PLATFORM_MSM8998)
+#if defined (PLATFORM_MSM8996) || (PLATFORM_MSM8998) || (PLATFORM_SDM845)
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
int hw_fd = -1;
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index edf5523..01a1c3f 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -162,6 +162,7 @@
struct voice_session *session = NULL;
int fd = 0;
int ret = 0;
+ bool is_voice_sess_active = false;
ALOGD("%s: enter:", __func__);
@@ -222,6 +223,12 @@
__func__, usecase_id);
} else {
session->state.current = session->state.new;
+
+ // The flag is not reset if another voice session is active as routing/mode is
+ // set globally instead of per session.
+ voice_extn_is_call_state_active(adev, &is_voice_sess_active);
+ if (!is_voice_sess_active)
+ adev->voice.in_call = false;
}
break;
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 1a2f2e8..c31b769 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msm8909 msm8998,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msm8909 msm8998 sdm845,$(TARGET_BOARD_PLATFORM)),)
LOCAL_PATH:= $(call my-dir)
@@ -43,7 +43,7 @@
################################################################################
-ifneq ($(filter msm8992 msm8994 msm8996 msm8909 msm8998,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8909 msm8998 sdm845,$(TARGET_BOARD_PLATFORM)),)
include $(CLEAR_VARS)