Merge "DO NOT MERGE: Merge Oreo MR1 into master" am: 4ff40ccbfd -s ours am: 690dbecf8a
am: f04cb090da -s ours
Change-Id: I739aa6ff31ad0412b839b6ed5521a815ad19cf06
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..5e5813f 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -4,13 +4,27 @@
include $(CLEAR_VARS)
+LOCAL_MODULE := libqcaudioperf
+
+LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)
+LOCAL_SHARED_LIBRARIES += libbase libhidlbase libhwbinder libutils android.hardware.power@1.2 liblog
+LOCAL_SRC_FILES := audio_perf.cpp
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_OWNER := qcom
+LOCAL_PROPRIETARY_MODULE := true
+
+include $(BUILD_SHARED_LIBRARY)
+
+include $(CLEAR_VARS)
+
LOCAL_ARM_MODE := arm
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
ifneq ($(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,$(TARGET_BOARD_PLATFORM)),)
@@ -44,6 +58,11 @@
ifneq ($(filter msm8998,$(TARGET_BOARD_PLATFORM)),)
LOCAL_CFLAGS := -DPLATFORM_MSM8998
LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+ MULTIPLE_HW_VARIANTS_ENABLED := true
+endif
+ifneq ($(filter sdm845,$(TARGET_BOARD_PLATFORM)),)
+ LOCAL_CFLAGS := -DPLATFORM_SDM845
+ LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
MULTIPLE_HW_VARIANTS_ENABLED := true
endif
@@ -144,12 +163,17 @@
LOCAL_SRC_FILES += audio_extn/spkr_protection.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_CIRRUS_SPKR_PROTECTION)),true)
+ LOCAL_CFLAGS += -DSPKR_PROT_ENABLED
+ LOCAL_SRC_FILES += audio_extn/cirrus_playback.c
+endif
+
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DSM_FEEDBACK)),true)
LOCAL_CFLAGS += -DDSM_FEEDBACK_ENABLED
LOCAL_SRC_FILES += audio_extn/dsm_feedback.c
endif
-ifneq ($(filter msm8992 msm8994 msm8996 msm8998,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msm8998 sdm845,$(TARGET_BOARD_PLATFORM)),)
# push codec/mad calibration to HW dep node
# applicable to msm8992/8994 or newer platforms
LOCAL_CFLAGS += -DHWDEP_CAL_ENABLED
@@ -161,6 +185,8 @@
LOCAL_SRC_FILES += audio_extn/sndmonitor.c
endif
+LOCAL_SHARED_LIBRARIES += libqcaudioperf
+
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
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/cirrus_playback.c b/hal/audio_extn/cirrus_playback.c
new file mode 100644
index 0000000..1f7a9f7
--- /dev/null
+++ b/hal/audio_extn/cirrus_playback.c
@@ -0,0 +1,541 @@
+/*
+ * Copyright (C) 2017 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_cirrus_playback"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+#include <fcntl.h>
+#include "../audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <sys/stat.h>
+#include <linux/types.h>
+#include <linux/ioctl.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <dlfcn.h>
+#include <math.h>
+#include <time.h>
+#include <unistd.h>
+#include <cutils/properties.h>
+#include "audio_extn.h"
+
+struct cirrus_playback_session {
+ void *adev_handle;
+ pthread_mutex_t mutex_fb_prot;
+ pthread_mutex_t mutex_pcm_stream;
+ pthread_t pcm_stream_thread;
+ struct pcm *pcm_rx;
+ struct pcm *pcm_tx;
+ bool spkr_prot_enable;
+};
+
+struct crus_sp_ioctl_header {
+ uint32_t size;
+ uint32_t module_id;
+ uint32_t param_id;
+ uint32_t data_length;
+ void *data;
+};
+
+/* Payload struct for getting calibration result from DSP module */
+struct cirrus_cal_result_t {
+ int32_t status_l;
+ int32_t checksum_l;
+ int32_t z_l;
+ int32_t status_r;
+ int32_t checksum_r;
+ int32_t z_r;
+ int32_t atemp;
+};
+
+/* Payload struct for setting the RX and TX use cases */
+struct crus_rx_run_case_ctrl_t {
+ int32_t value;
+ int32_t status_l;
+ int32_t checksum_l;
+ int32_t z_l;
+ int32_t status_r;
+ int32_t checksum_r;
+ int32_t z_r;
+ int32_t atemp;
+};
+
+#define CRUS_SP_FILE "/dev/msm_cirrus_playback"
+#define CRUS_CAL_FILE "/persist/audio/audio.cal"
+
+#define CRUS_SP_USECASE_MIXER "Cirrus SP Usecase Config"
+#define CRUS_SP_EXT_CONFIG_MIXER "Cirrus SP EXT Config"
+
+#define CIRRUS_SP 0x10027053
+
+#define CRUS_MODULE_ID_TX 0x00000002
+#define CRUS_MODULE_ID_RX 0x00000001
+
+#define CRUS_PARAM_RX_SET_USECASE 0x00A1AF02
+#define CRUS_PARAM_TX_SET_USECASE 0x00A1BF0A
+
+#define CRUS_PARAM_RX_SET_CALIB 0x00A1AF03
+#define CRUS_PARAM_TX_SET_CALIB 0x00A1BF03
+
+#define CRUS_PARAM_RX_SET_EXT_CONFIG 0x00A1AF05
+#define CRUS_PARAM_TX_SET_EXT_CONFIG 0x00A1BF08
+
+#define CRUS_PARAM_RX_GET_TEMP 0x00A1AF07
+#define CRUS_PARAM_TX_GET_TEMP_CAL 0x00A1BF06
+// variables based on CSPL tuning file, max parameter length is 96 integers (384 bytes)
+#define CRUS_PARAM_TEMP_MAX_LENGTH 384
+
+#define CRUS_AFE_PARAM_ID_ENABLE 0x00010203
+
+#define CRUS_SP_IOCTL_MAGIC 'a'
+
+#define CRUS_SP_IOCTL_GET _IOWR(CRUS_SP_IOCTL_MAGIC, 219, void *)
+#define CRUS_SP_IOCTL_SET _IOWR(CRUS_SP_IOCTL_MAGIC, 220, void *)
+#define CRUS_SP_IOCTL_GET_CALIB _IOWR(CRUS_SP_IOCTL_MAGIC, 221, void *)
+#define CRUS_SP_IOCTL_SET_CALIB _IOWR(CRUS_SP_IOCTL_MAGIC, 222, void *)
+
+#define CRUS_SP_DEFAULT_AMBIENT_TEMP 23
+
+static struct pcm_config pcm_config_cirrus_tx = {
+ .channels = 2,
+ .rate = 48000,
+ .period_size = 320,
+ .period_count = 4,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+ .stop_threshold = INT_MAX,
+ .avail_min = 0,
+};
+
+static struct pcm_config pcm_config_cirrus_rx = {
+ .channels = 2,
+ .rate = 48000,
+ .period_size = 320,
+ .period_count = 4,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+ .stop_threshold = INT_MAX,
+ .avail_min = 0,
+};
+
+static struct cirrus_playback_session handle;
+
+static void *audio_extn_cirrus_run_calibration() {
+ struct audio_device *adev = handle.adev_handle;
+ struct crus_sp_ioctl_header header;
+ struct cirrus_cal_result_t result;
+ struct mixer_ctl *ctl;
+ FILE *cal_file;
+ int ret = 0, dev_file;
+ char *buffer = NULL;
+ uint32_t option = 1;
+ struct timespec timeout;
+
+ ALOGI("%s: Calibration thread", __func__);
+
+ timeout.tv_sec = 10;
+ timeout.tv_nsec = 0;
+
+ dev_file = open(CRUS_SP_FILE, O_RDWR | O_NONBLOCK);
+ if (dev_file < 0) {
+ ALOGE("%s: Failed to open Cirrus Playback IOCTL (%d)",
+ __func__, dev_file);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ buffer = calloc(1, CRUS_PARAM_TEMP_MAX_LENGTH);
+ if (!buffer) {
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ cal_file = fopen(CRUS_CAL_FILE, "r");
+ if (cal_file) {
+ size_t bytes;
+ bytes = fread(&result, 1, sizeof(result), cal_file);
+ if (bytes < sizeof(result)) {
+ ALOGE("%s: Cirrus SP calibration file cannot be read (%d)",
+ __func__, ret);
+ ret = -EINVAL;
+ fclose(cal_file);
+ goto exit;
+ }
+
+ fclose(cal_file);
+ } else {
+
+ ALOGV("%s: Calibrating...", __func__);
+
+ header.size = sizeof(header);
+ header.module_id = CRUS_MODULE_ID_RX;
+ header.param_id = CRUS_PARAM_RX_SET_CALIB;
+ header.data_length = sizeof(option);
+ header.data = &option;
+
+ ret = ioctl(dev_file, CRUS_SP_IOCTL_SET, &header);
+ if (ret < 0) {
+ ALOGE("%s: Cirrus SP calibration IOCTL failure (%d)",
+ __func__, ret);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ header.size = sizeof(header);
+ header.module_id = CRUS_MODULE_ID_TX;
+ header.param_id = CRUS_PARAM_TX_SET_CALIB;
+ header.data_length = sizeof(option);
+ header.data = &option;
+
+ ret = ioctl(dev_file, CRUS_SP_IOCTL_SET, &header);
+ if (ret < 0) {
+ ALOGE("%s: Cirrus SP calibration IOCTL failure (%d)",
+ __func__, ret);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ sleep(2);
+
+ header.size = sizeof(header);
+ header.module_id = CRUS_MODULE_ID_TX;
+ header.param_id = CRUS_PARAM_TX_GET_TEMP_CAL;
+ header.data_length = sizeof(result);
+ header.data = &result;
+
+ ret = ioctl(dev_file, CRUS_SP_IOCTL_GET, &header);
+ if (ret < 0) {
+ ALOGE("%s: Cirrus SP calibration IOCTL failure (%d)",
+ __func__, ret);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ result.atemp = CRUS_SP_DEFAULT_AMBIENT_TEMP;
+
+ // TODO: Save calibrated data to file
+ }
+
+ header.size = sizeof(header);
+ header.module_id = CRUS_MODULE_ID_TX;
+ header.param_id = 0;
+ header.data_length = sizeof(result);
+ header.data = &result;
+
+ ret = ioctl(dev_file, CRUS_SP_IOCTL_SET_CALIB, &header);
+
+ if (ret < 0) {
+ ALOGE("%s: Cirrus SP calibration IOCTL failure (%d)", __func__, ret);
+ close(dev_file);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ CRUS_SP_USECASE_MIXER);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, CRUS_SP_USECASE_MIXER);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ mixer_ctl_set_value(ctl, 0, 0); // Set RX external firmware config
+
+ sleep(1);
+
+ header.size = sizeof(header);
+ header.module_id = CRUS_MODULE_ID_RX;
+ header.param_id = CRUS_PARAM_RX_GET_TEMP;
+ header.data_length = sizeof(buffer);
+ header.data = buffer;
+
+ ret = ioctl(dev_file, CRUS_SP_IOCTL_GET, &header);
+ if (ret < 0) {
+ ALOGE("%s: Cirrus SP temperature IOCTL failure (%d)", __func__, ret);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ ALOGI("%s: Cirrus SP successfully calibrated", __func__);
+
+exit:
+ close(dev_file);
+ free(buffer);
+ ALOGV("%s: Exit", __func__);
+
+ return NULL;
+}
+
+
+static void *audio_extn_cirrus_pcm_stream_thread() {
+ struct audio_device *adev = handle.adev_handle;
+ struct audio_usecase *uc_info_rx = NULL;
+ int ret = 0;
+ int32_t pcm_dev_rx_id = 0;
+ uint32_t rx_use_case = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
+ uint32_t retries = 5;
+
+ pthread_mutex_lock(&handle.mutex_pcm_stream);
+
+ ALOGI("%s: PCM Stream thread", __func__);
+
+ while (!adev->platform && retries) {
+ sleep(1);
+ ALOGI("%s: Waiting...", __func__);
+ retries--;
+ }
+
+ uc_info_rx = (struct audio_usecase *)calloc(1, sizeof(*uc_info_rx));
+ if (!uc_info_rx) {
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ uc_info_rx->id = rx_use_case;
+ uc_info_rx->type = PCM_PLAYBACK;
+ uc_info_rx->in_snd_device = SND_DEVICE_NONE;
+ uc_info_rx->stream.out = adev->primary_output;
+ uc_info_rx->out_snd_device = SND_DEVICE_OUT_SPEAKER;
+ list_add_tail(&adev->usecase_list, &uc_info_rx->list);
+
+ enable_snd_device(adev, SND_DEVICE_OUT_SPEAKER);
+ enable_audio_route(adev, uc_info_rx);
+
+ handle.pcm_rx = pcm_open(adev->snd_card, pcm_dev_rx_id,
+ PCM_OUT, &pcm_config_cirrus_rx);
+
+ if (!handle.pcm_rx || !pcm_is_ready(handle.pcm_rx)) {
+ ALOGE("%s: PCM device not ready: %s", __func__,
+ pcm_get_error(handle.pcm_rx ? handle.pcm_rx : 0));
+ ret = -EINVAL;
+ goto close_stream;
+ }
+
+ if (pcm_start(handle.pcm_rx) < 0) {
+ ALOGE("%s: pcm start for RX failed; error = %s", __func__,
+ pcm_get_error(handle.pcm_rx));
+ ret = -EINVAL;
+ goto close_stream;
+ }
+
+ ALOGV("%s: PCM thread streaming", __func__);
+
+ audio_extn_cirrus_run_calibration();
+
+close_stream:
+ if (handle.pcm_rx) {
+ ALOGV("%s: pcm_rx_close", __func__);
+ pcm_close(handle.pcm_rx);
+ handle.pcm_rx = NULL;
+ }
+
+ disable_audio_route(adev, uc_info_rx);
+ disable_snd_device(adev, SND_DEVICE_OUT_SPEAKER);
+ list_remove(&uc_info_rx->list);
+ free(uc_info_rx);
+
+exit:
+ pthread_mutex_unlock(&handle.mutex_pcm_stream);
+ ALOGV("%s: Exit", __func__);
+
+ pthread_exit(0);
+ return NULL;
+}
+
+void audio_extn_spkr_prot_init(void *adev) {
+ ALOGI("%s: Initialize Cirrus Logic Playback module", __func__);
+
+ struct snd_card_split *snd_split_handle = NULL;
+
+ if (!adev) {
+ ALOGE("%s: Invalid params", __func__);
+ return;
+ }
+
+ memset(&handle, 0, sizeof(handle));
+
+ snd_split_handle = audio_extn_get_snd_card_split();
+
+ /* FIXME: REMOVE THIS AFTER B1C1 P1.0 SUPPORT */
+ if (!strcmp(snd_split_handle->form_factor, "tdm")) {
+ handle.spkr_prot_enable = true;
+ } else {
+ handle.spkr_prot_enable = false;
+ }
+
+ handle.adev_handle = adev;
+
+ pthread_mutex_init(&handle.mutex_fb_prot, NULL);
+ pthread_mutex_init(&handle.mutex_pcm_stream, NULL);
+
+ (void)pthread_create(&handle.pcm_stream_thread,
+ (const pthread_attr_t *) NULL,
+ audio_extn_cirrus_pcm_stream_thread, &handle);
+}
+
+int audio_extn_spkr_prot_start_processing(snd_device_t snd_device) {
+ struct audio_usecase *uc_info_tx;
+ struct audio_device *adev = handle.adev_handle;
+ int32_t pcm_dev_tx_id = -1, ret = 0;
+
+ ALOGV("%s: Entry", __func__);
+
+ if (!adev) {
+ ALOGE("%s: Invalid params", __func__);
+ return -EINVAL;
+ }
+
+ uc_info_tx = (struct audio_usecase *)calloc(1, sizeof(*uc_info_tx));
+ if (!uc_info_tx) {
+ return -ENOMEM;
+ }
+
+ audio_route_apply_and_update_path(adev->audio_route,
+ platform_get_snd_device_name(snd_device));
+
+ pthread_mutex_lock(&handle.mutex_fb_prot);
+ uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
+ uc_info_tx->type = PCM_CAPTURE;
+ uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ uc_info_tx->out_snd_device = SND_DEVICE_NONE;
+ handle.pcm_tx = NULL;
+
+ list_add_tail(&adev->usecase_list, &uc_info_tx->list);
+
+ enable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ enable_audio_route(adev, uc_info_tx);
+
+ pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
+
+ if (pcm_dev_tx_id < 0) {
+ ALOGE("%s: Invalid pcm device for usecase (%d)",
+ __func__, uc_info_tx->id);
+ ret = -ENODEV;
+ goto exit;
+ }
+
+ handle.pcm_tx = pcm_open(adev->snd_card,
+ pcm_dev_tx_id,
+ PCM_IN, &pcm_config_cirrus_tx);
+
+ if (!handle.pcm_tx || !pcm_is_ready(handle.pcm_tx)) {
+ ALOGD("%s: PCM device not ready: %s", __func__,
+ pcm_get_error(handle.pcm_tx ? handle.pcm_tx : 0));
+ ret = -EIO;
+ goto exit;
+ }
+
+ if (pcm_start(handle.pcm_tx) < 0) {
+ ALOGI("%s: retrying pcm_start...", __func__);
+ usleep(500 * 1000);
+ if (pcm_start(handle.pcm_tx) < 0) {
+ ALOGI("%s: pcm start for TX failed; error = %s", __func__,
+ pcm_get_error(handle.pcm_tx));
+ ret = -EINVAL;
+ }
+ }
+
+exit:
+ if (ret) {
+ ALOGI("%s: Disable and bail out", __func__);
+ if (handle.pcm_tx) {
+ ALOGI("%s: pcm_tx_close", __func__);
+ pcm_close(handle.pcm_tx);
+ handle.pcm_tx = NULL;
+ }
+
+ disable_audio_route(adev, uc_info_tx);
+ disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ list_remove(&uc_info_tx->list);
+ free(uc_info_tx);
+ }
+
+ pthread_mutex_unlock(&handle.mutex_fb_prot);
+ ALOGV("%s: Exit", __func__);
+ return ret;
+}
+
+void audio_extn_spkr_prot_stop_processing(snd_device_t snd_device) {
+ struct audio_usecase *uc_info_tx;
+ struct audio_device *adev = handle.adev_handle;
+
+ ALOGV("%s: Entry", __func__);
+
+ pthread_mutex_lock(&handle.mutex_fb_prot);
+
+ uc_info_tx = get_usecase_from_list(adev, USECASE_AUDIO_SPKR_CALIB_TX);
+
+ if (uc_info_tx) {
+ if (handle.pcm_tx) {
+ ALOGI("%s: pcm_tx_close", __func__);
+ pcm_close(handle.pcm_tx);
+ handle.pcm_tx = NULL;
+ }
+
+ disable_audio_route(adev, uc_info_tx);
+ disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ list_remove(&uc_info_tx->list);
+ free(uc_info_tx);
+
+ audio_route_reset_path(adev->audio_route,
+ platform_get_snd_device_name(snd_device));
+ }
+
+ pthread_mutex_unlock(&handle.mutex_fb_prot);
+
+ ALOGV("%s: Exit", __func__);
+}
+
+bool audio_extn_spkr_prot_is_enabled() {
+ return handle.spkr_prot_enable;
+}
+
+int audio_extn_spkr_prot_get_acdb_id(snd_device_t snd_device) {
+ int acdb_id;
+
+ switch (snd_device) {
+ case SND_DEVICE_OUT_SPEAKER:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED);
+ break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
+ break;
+ default:
+ acdb_id = -EINVAL;
+ break;
+ }
+ return acdb_id;
+}
+
+int audio_extn_get_spkr_prot_snd_device(snd_device_t snd_device) {
+ switch(snd_device) {
+ case SND_DEVICE_OUT_SPEAKER:
+ return SND_DEVICE_OUT_SPEAKER_PROTECTED;
+ case SND_DEVICE_OUT_VOICE_SPEAKER:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ default:
+ return snd_device;
+ }
+}
+
+void audio_extn_spkr_prot_calib_cancel(__unused void *adev) {
+ // FIXME: wait or cancel audio_extn_cirrus_run_calibration
+}
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..85c886b 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -51,6 +51,7 @@
#include <audio_utils/clock.h>
#include "audio_hw.h"
#include "audio_extn.h"
+#include "audio_perf.h"
#include "platform_api.h"
#include <platform.h>
#include "voice_extn.h"
@@ -999,9 +1000,9 @@
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,
@@ -1460,6 +1461,7 @@
list_add_tail(&adev->usecase_list, &uc_info->list);
+ audio_streaming_hint_start();
audio_extn_perf_lock_acquire();
select_devices(adev, in->usecase);
@@ -1526,6 +1528,7 @@
}
}
register_in_stream(in);
+ audio_streaming_hint_end();
audio_extn_perf_lock_release();
ALOGV("%s: exit", __func__);
@@ -1533,6 +1536,7 @@
error_open:
stop_input_stream(in);
+ audio_streaming_hint_end();
audio_extn_perf_lock_release();
error_config:
@@ -1799,6 +1803,9 @@
adev->visualizer_stop_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_stop_output != NULL)
adev->offload_effects_stop_output(out->handle, out->pcm_device_id);
+ } else if (out->usecase == USECASE_AUDIO_PLAYBACK_ULL ||
+ out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
+ audio_low_latency_hint_end();
}
/* 1. Get and set stream specific mixer controls */
@@ -1867,6 +1874,7 @@
list_add_tail(&adev->usecase_list, &uc_info->list);
+ audio_streaming_hint_start();
audio_extn_perf_lock_acquire();
select_devices(adev, out->usecase);
@@ -1953,9 +1961,15 @@
}
}
register_out_stream(out);
+ audio_streaming_hint_end();
audio_extn_perf_lock_release();
audio_extn_tfa_98xx_enable_speaker();
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_ULL ||
+ out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
+ audio_low_latency_hint_start();
+ }
+
// consider a scenario where on pause lower layers are tear down.
// so on resume, swap mixer control need to be sent only when
// backend is active, hence rather than sending from enable device
@@ -1966,6 +1980,7 @@
ALOGV("%s: exit", __func__);
return 0;
error_open:
+ audio_streaming_hint_end();
audio_extn_perf_lock_release();
stop_output_stream(out);
error_config:
@@ -3680,18 +3695,17 @@
{
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 +3715,14 @@
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 +3744,38 @@
&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 +3798,20 @@
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 +3823,11 @@
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 +3840,136 @@
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 +3980,38 @@
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 +4140,7 @@
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 +4686,9 @@
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 +4856,7 @@
}
if (adev->adm_deinit)
adev->adm_deinit(adev->adm_data);
+ pthread_mutex_destroy(&adev->lock);
free(device);
}
diff --git a/hal/audio_perf.cpp b/hal/audio_perf.cpp
new file mode 100644
index 0000000..671a324
--- /dev/null
+++ b/hal/audio_perf.cpp
@@ -0,0 +1,116 @@
+/*
+ * Copyright (C) 2017 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_primary"
+
+#include <cinttypes>
+
+#include <utils/Log.h>
+#include <utils/Mutex.h>
+
+#include <android/hardware/power/1.2/IPower.h>
+
+#include "audio_perf.h"
+
+using android::hardware::power::V1_2::IPower;
+using android::hardware::power::V1_2::PowerHint;
+using android::hardware::power::V1_2::toString;
+using android::hardware::Return;
+using android::hardware::Void;
+using android::hardware::hidl_death_recipient;
+using android::hidl::base::V1_0::IBase;
+
+// Do not use gPowerHAL, use getPowerHal to retrieve a copy instead
+static android::sp<IPower> gPowerHal_ = nullptr;
+// Protect gPowerHal_
+static std::mutex gPowerHalMutex;
+
+// PowerHalDeathRecipient to invalid the client when service dies
+struct PowerHalDeathRecipient : virtual public hidl_death_recipient {
+ // hidl_death_recipient interface
+ virtual void serviceDied(uint64_t, const android::wp<IBase>&) override {
+ std::lock_guard<std::mutex> lock(gPowerHalMutex);
+ ALOGE("PowerHAL just died");
+ gPowerHal_ = nullptr;
+ }
+};
+
+// Retrieve a copy of client
+static android::sp<IPower> getPowerHal() {
+ std::lock_guard<std::mutex> lock(gPowerHalMutex);
+ static android::sp<PowerHalDeathRecipient> gPowerHalDeathRecipient = nullptr;
+ static bool gPowerHalExists = true;
+
+ if (gPowerHalExists && gPowerHal_ == nullptr) {
+ gPowerHal_ = IPower::getService();
+
+ if (gPowerHal_ == nullptr) {
+ ALOGE("Unable to get Power service");
+ gPowerHalExists = false;
+ } else {
+ if (gPowerHalDeathRecipient == nullptr) {
+ gPowerHalDeathRecipient = new PowerHalDeathRecipient();
+ }
+ Return<bool> linked = gPowerHal_->linkToDeath(
+ gPowerHalDeathRecipient, 0 /* cookie */);
+ if (!linked.isOk()) {
+ ALOGE("Transaction error in linking to PowerHAL death: %s",
+ linked.description().c_str());
+ gPowerHal_ = nullptr;
+ } else if (!linked) {
+ ALOGW("Unable to link to PowerHal death notifications");
+ gPowerHal_ = nullptr;
+ } else {
+ ALOGD("Connect to PowerHAL and link to death "
+ "notification successfully");
+ }
+ }
+ }
+ return gPowerHal_;
+}
+
+static bool powerHint(PowerHint hint, int32_t data) {
+ android::sp<IPower> powerHal = getPowerHal();
+ if (powerHal == nullptr) {
+ return false;
+ }
+
+ auto ret = powerHal->powerHintAsync_1_2(hint, data);
+
+ if (!ret.isOk()) {
+ ALOGE("powerHint failed, hint: %s, data: %" PRId32 ", error: %s",
+ toString(hint).c_str(),
+ data,
+ ret.description().c_str());
+ }
+ return ret.isOk();
+}
+
+int audio_streaming_hint_start() {
+ return powerHint(PowerHint::AUDIO_STREAMING, 1);
+}
+
+int audio_streaming_hint_end() {
+ return powerHint(PowerHint::AUDIO_STREAMING, 0);
+}
+
+int audio_low_latency_hint_start() {
+ return powerHint(PowerHint::AUDIO_LOW_LATENCY, 1);
+}
+
+int audio_low_latency_hint_end() {
+ return powerHint(PowerHint::AUDIO_LOW_LATENCY, 0);
+}
diff --git a/hal/audio_perf.h b/hal/audio_perf.h
new file mode 100644
index 0000000..4263086
--- /dev/null
+++ b/hal/audio_perf.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2017 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.
+ */
+
+#ifndef __QAUDIOPERF_H__
+#define __QAUDIOPERF_H__
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+int audio_streaming_hint_start();
+int audio_streaming_hint_end();
+
+int audio_low_latency_hint_start();
+int audio_low_latency_hint_end();
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif //__QCAMREAPERF_H__
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)