Merge "hal: Fix calibration for external speakers."
diff --git a/Android.mk b/Android.mk
index 6d0ec5f..6e129a1 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916 msm8994 msm8992 msm8909 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916 msm8994 msm8992 msm8909 msm8996 msm8952,$(TARGET_BOARD_PLATFORM)),)
MY_LOCAL_PATH := $(call my-dir)
diff --git a/hal/Android.mk b/hal/Android.mk
index c0cff58..99b6b39 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -32,7 +32,7 @@
endif
endif
-ifneq ($(filter msm8916 msm8909,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8916 msm8909 msm8952,$(TARGET_BOARD_PLATFORM)),)
AUDIO_PLATFORM = msm8916
MULTIPLE_HW_VARIANTS_ENABLED := true
LOCAL_CFLAGS := -DPLATFORM_MSM8916
@@ -194,6 +194,11 @@
LOCAL_CFLAGS += -DHDMI_PASSTHROUGH_ENABLED
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SOURCE_TRACKING)),true)
+ LOCAL_CFLAGS += -DSOURCE_TRACKING_ENABLED
+ LOCAL_SRC_FILES += audio_extn/source_track.c
+endif
+
LOCAL_SHARED_LIBRARIES := \
liblog \
libcutils \
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index ff485e5..688e265 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -91,6 +91,17 @@
struct str_parms *parms);
#endif
+#ifndef SOURCE_TRACKING_ENABLED
+#define audio_extn_source_track_set_parameters(adev, parms) (0)
+#define audio_extn_source_track_get_parameters(adev, query, reply) (0)
+#else
+void audio_extn_source_track_set_parameters(struct audio_device *adev,
+ struct str_parms *parms);
+void audio_extn_source_track_get_parameters(struct audio_device *adev,
+ struct str_parms *query,
+ struct str_parms *reply);
+#endif
+
#ifndef CUSTOM_STEREO_ENABLED
#define audio_extn_customstereo_set_parameters(adev, parms) (0)
#else
@@ -332,7 +343,7 @@
int channel_count)
{
struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "Playback Channel Map";
+ const char *mixer_ctl_name = "Playback Device Channel Map";
int set_values[8] = {0};
int ret;
ALOGV("%s channel_count:%d",__func__, channel_count);
@@ -535,6 +546,7 @@
audio_extn_customstereo_set_parameters(adev, parms);
audio_extn_hpx_set_parameters(adev, parms);
audio_extn_pm_set_parameters(parms);
+ audio_extn_source_track_set_parameters(adev, parms);
}
void audio_extn_get_parameters(const struct audio_device *adev,
@@ -547,6 +559,7 @@
get_active_offload_usecases(adev, query, reply);
audio_extn_dts_eagle_get_parameters(adev, query, reply);
audio_extn_hpx_get_parameters(query, reply);
+ audio_extn_source_track_get_parameters(adev, query, reply);
kv_pairs = str_parms_to_str(reply);
ALOGD_IF(kv_pairs != NULL, "%s: returns %s", __func__, kv_pairs);
diff --git a/hal/audio_extn/dts_eagle.c b/hal/audio_extn/dts_eagle.c
index 83c3506..52d7abb 100644
--- a/hal/audio_extn/dts_eagle.c
+++ b/hal/audio_extn/dts_eagle.c
@@ -150,20 +150,16 @@
if (strncmp("true", prop, sizeof("true")))
return;
int fd, n = 0;
- if ((fd = open(FADE_NOTIFY_FILE, O_RDONLY)) < 0)
- ALOGV("No fade node");
- else {
- ALOGV("fade node exists, remove it before creating it");
- close(fd);
- remove(FADE_NOTIFY_FILE);
+ if ((fd = open(FADE_NOTIFY_FILE, O_TRUNC|O_WRONLY)) < 0) {
+ ALOGV("No fade node, create one");
+ fd = creat(FADE_NOTIFY_FILE, S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH);
+ if (fd < 0) {
+ ALOGE("DTS_EAGLE_HAL (%s): Creating fade notifier node failed", __func__);
+ return;
+ }
+ chmod(FADE_NOTIFY_FILE, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
}
- fd = creat(FADE_NOTIFY_FILE, S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH);
char *str = need_data ? "need" : "have";
- if (fd < 0) {
- ALOGE("DTS_EAGLE_HAL (%s): opening fade notifier node failed", __func__);
- return;
- }
- chmod(FADE_NOTIFY_FILE, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
n = write(fd, str, strlen(str));
close(fd);
if (n > 0)
diff --git a/hal/audio_extn/source_track.c b/hal/audio_extn/source_track.c
new file mode 100644
index 0000000..316e52d
--- /dev/null
+++ b/hal/audio_extn/source_track.c
@@ -0,0 +1,640 @@
+/*
+ * Copyright (c) 2015, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+#define LOG_TAG "source_track"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include "voice_extn.h"
+#include <stdlib.h>
+#include <cutils/str_parms.h>
+
+#ifdef SOURCE_TRACKING_ENABLED
+/* Audio Paramater Key to identify the list of start angles.
+ * Starting angle (in degrees) defines the boundary starting angle for each sector.
+ */
+#define AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES "SoundFocus.start_angles"
+/* Audio Paramater Key to identify the list of enable flags corresponding to each sector.
+ */
+#define AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS "SoundFocus.enable_sectors"
+/* Audio Paramater Key to identify the gain step value to be applied to all enabled sectors.
+ */
+#define AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP "SoundFocus.gain_step"
+/* Audio Paramater Key to identify the list of voice activity detector outputs corresponding
+ * to each sector.
+ */
+#define AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD "SourceTrack.vad"
+/* Audio Paramater Key to identify the direction (in degrees) of arrival for desired talker
+ * (dominant source of speech).
+ */
+#define AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH "SourceTrack.doa_speech"
+/* Audio Paramater Key to identify the list of directions (in degrees) of arrival for
+ * interferers (interfering noise sources).
+ */
+#define AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE "SourceTrack.doa_noise"
+/* Audio Paramater Key to identify the list of sound strength indicators at each degree
+ * of the horizontal plane referred to by a full circle (360 degrees).
+ */
+#define AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY "SourceTrack.polar_activity"
+
+#define BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES 0x1
+#define BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS 0x2
+#define BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP 0x4
+#define BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD 0x8
+#define BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH 0x10
+#define BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE 0x20
+#define BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY 0x40
+
+#define BITMASK_AUDIO_PARAMETER_KEYS_SOUND_FOCUS \
+ (BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES |\
+ BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS |\
+ BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP)
+
+#define BITMASK_AUDIO_PARAMETER_KEYS_SOURCE_TRACKING \
+ (BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD |\
+ BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH |\
+ BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE |\
+ BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY)
+
+#define MAX_SECTORS 8
+#define MAX_STR_SIZE 2048
+
+struct audio_device_to_audio_interface audio_device_to_interface_table[];
+int audio_device_to_interface_table_len;
+
+struct sound_focus_param {
+ uint16_t start_angle[MAX_SECTORS];
+ uint8_t enable[MAX_SECTORS];
+ uint16_t gain_step;
+};
+
+struct source_tracking_param {
+ uint8_t vad[MAX_SECTORS];
+ uint16_t doa_speech;
+ uint16_t doa_noise[3];
+ uint8_t polar_activity[360];
+};
+
+static int add_audio_intf_name_to_mixer_ctl(audio_devices_t device, char *mixer_ctl_name,
+ struct audio_device_to_audio_interface *table, int len)
+{
+ int ret = 0;
+ int i;
+
+ if (table == NULL) {
+ ALOGE("%s: table is NULL", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (mixer_ctl_name == NULL) {
+ ret = -EINVAL;
+ goto done;
+ }
+
+ for (i=0; i < len; i++) {
+ if (device == table[i].device) {
+ strlcat(mixer_ctl_name, " ", MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_ctl_name, table[i].interface_name, MIXER_PATH_MAX_LENGTH);
+ break;
+ }
+ }
+
+ if (i == len) {
+ ALOGE("%s: Audio Device not found in the table", __func__);
+
+ ret = -EINVAL;
+ }
+done:
+ return ret;
+}
+
+static bool is_stt_supported_snd_device(snd_device_t snd_device)
+{
+ bool ret = false;
+
+ switch (snd_device) {
+ case SND_DEVICE_IN_HANDSET_DMIC:
+ case SND_DEVICE_IN_HANDSET_DMIC_AEC:
+ case SND_DEVICE_IN_HANDSET_DMIC_NS:
+ case SND_DEVICE_IN_HANDSET_DMIC_AEC_NS:
+ case SND_DEVICE_IN_HANDSET_STEREO_DMIC:
+ case SND_DEVICE_IN_HANDSET_QMIC:
+ case SND_DEVICE_IN_VOICE_DMIC:
+ case SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE:
+ case SND_DEVICE_IN_HEADSET_MIC_FLUENCE:
+ case SND_DEVICE_IN_SPEAKER_DMIC:
+ case SND_DEVICE_IN_SPEAKER_DMIC_AEC:
+ case SND_DEVICE_IN_SPEAKER_DMIC_NS:
+ case SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS:
+ case SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS_BROADSIDE:
+ case SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE:
+ case SND_DEVICE_IN_SPEAKER_DMIC_NS_BROADSIDE:
+ case SND_DEVICE_IN_SPEAKER_QMIC_AEC:
+ case SND_DEVICE_IN_SPEAKER_QMIC_NS:
+ case SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS:
+ case SND_DEVICE_IN_VOICE_SPEAKER_DMIC:
+ case SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE:
+ case SND_DEVICE_IN_VOICE_SPEAKER_QMIC:
+ ret = true;
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+audio_devices_t get_input_audio_device(audio_devices_t device)
+{
+ audio_devices_t in_device = device;
+
+ switch (device) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ in_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ break;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ in_device = AUDIO_DEVICE_IN_BACK_MIC;
+ break;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ in_device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ break;
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ in_device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ break;
+ default:
+ break;
+ }
+
+ return in_device;
+}
+
+static int derive_mixer_ctl_from_usecase_intf(struct audio_device *adev,
+ char *mixer_ctl_name) {
+ struct audio_usecase *usecase = NULL;
+ audio_devices_t in_device;
+ int ret = 0;
+
+ if (mixer_ctl_name == NULL) {
+ ALOGE("%s: mixer_ctl_name is NULL", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (voice_is_in_call(adev)) {
+ strlcat(mixer_ctl_name, " ", MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_ctl_name, "Voice Tx", MIXER_PATH_MAX_LENGTH);
+ usecase = get_usecase_from_list(adev,
+ get_usecase_id_from_usecase_type(adev, VOICE_CALL));
+ } else if (voice_extn_compress_voip_is_active(adev)) {
+ strlcat(mixer_ctl_name, " ", MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_ctl_name, "Voice Tx", MIXER_PATH_MAX_LENGTH);
+ usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
+ } else {
+ strlcat(mixer_ctl_name, " ", MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_ctl_name, "Audio Tx", MIXER_PATH_MAX_LENGTH);
+ usecase = get_usecase_from_list(adev, get_usecase_id_from_usecase_type(adev, PCM_CAPTURE));
+ }
+
+ if (usecase && (usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
+ if (is_stt_supported_snd_device(usecase->in_snd_device)) {
+ in_device = get_input_audio_device(usecase->devices);
+ ret = add_audio_intf_name_to_mixer_ctl(in_device, mixer_ctl_name,
+ audio_device_to_interface_table, audio_device_to_interface_table_len);
+ } else {
+ ALOGE("%s: Sound Focus/Source Tracking not supported on the input sound device (%s)",
+ __func__, platform_get_snd_device_name(usecase->in_snd_device));
+
+ ret = -EINVAL;
+ }
+ } else {
+ ALOGE("%s: No use case is active which supports Sound Focus/Source Tracking",
+ __func__);
+
+ ret = -EINVAL;
+ }
+
+done:
+ return ret;
+}
+
+static int parse_soundfocus_sourcetracking_keys(struct str_parms *parms)
+{
+ char *str;
+ char *value = NULL;
+ int val, len;
+ int ret = 0, err;
+ char *kv_pairs = str_parms_to_str(parms);
+
+ ALOGV_IF(kv_pairs != NULL, "%s: enter: %s", __func__, kv_pairs);
+
+ len = strlen(kv_pairs);
+ value = (char*)calloc(len, sizeof(char));
+ if(value == NULL) {
+ ret = -ENOMEM;
+ ALOGE("%s: failed to allocate memory", __func__);
+
+ goto done;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE;
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY,
+ value, len);
+ if (err >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY);
+ ret = ret | BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY;
+ }
+
+done:
+ free(kv_pairs);
+ if(value != NULL)
+ free(value);
+ ALOGV("%s: returning bitmask = %d", __func__, ret);
+
+ return ret;
+}
+
+static int get_soundfocus_sourcetracking_data(struct audio_device *adev,
+ const int bitmask,
+ struct sound_focus_param *sound_focus_data,
+ struct source_tracking_param *source_tracking_data)
+{
+ struct mixer_ctl *ctl;
+ char sound_focus_mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = "Sound Focus";
+ char source_tracking_mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = "Source Tracking";
+ int ret = -EINVAL;
+ int i, count;
+
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEYS_SOUND_FOCUS) {
+ /* Derive the mixer control name based on the use case and the audio interface
+ * for the corresponding audio device
+ */
+ ret = derive_mixer_ctl_from_usecase_intf(adev, sound_focus_mixer_ctl_name);
+ if (ret != 0) {
+ ALOGE("%s: Could not get Sound Focus Params", __func__);
+
+ goto done;
+ } else {
+ ALOGV("%s: Mixer Ctl name: %s", __func__, sound_focus_mixer_ctl_name);
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, sound_focus_mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, sound_focus_mixer_ctl_name);
+
+ ret = -EINVAL;
+ goto done;
+ } else {
+ ALOGV("%s: Getting Sound Focus Params", __func__);
+
+ mixer_ctl_update(ctl);
+ count = mixer_ctl_get_num_values(ctl);
+ if (count != sizeof(struct sound_focus_param)) {
+ ALOGE("%s: mixer_ctl_get_num_values() invalid sound focus data size", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ret = mixer_ctl_get_array(ctl, (void *)sound_focus_data, count);
+ if (ret != 0) {
+ ALOGE("%s: mixer_ctl_get_array() failed to get Sound Focus Params", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+ }
+ }
+
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEYS_SOURCE_TRACKING) {
+ /* Derive the mixer control name based on the use case and the audio interface
+ * for the corresponding audio device
+ */
+ ret = derive_mixer_ctl_from_usecase_intf(adev, source_tracking_mixer_ctl_name);
+ if (ret != 0) {
+ ALOGE("%s: Could not get Source Tracking Params", __func__);
+
+ goto done;
+ } else {
+ ALOGV("%s: Mixer Ctl name: %s", __func__, source_tracking_mixer_ctl_name);
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, source_tracking_mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, source_tracking_mixer_ctl_name);
+
+ ret = -EINVAL;
+ goto done;
+ } else {
+ ALOGV("%s: Getting Source Tracking Params", __func__);
+
+ mixer_ctl_update(ctl);
+ count = mixer_ctl_get_num_values(ctl);
+ if (count != sizeof(struct source_tracking_param)) {
+ ALOGE("%s: mixer_ctl_get_num_values() invalid source tracking data size", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ret = mixer_ctl_get_array(ctl, (void *)source_tracking_data, count);
+ if (ret != 0) {
+ ALOGE("%s: mixer_ctl_get_array() failed to get Source Tracking Params", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+ }
+ }
+
+done:
+ return ret;
+}
+
+static void send_soundfocus_sourcetracking_params(struct str_parms *reply,
+ const int bitmask,
+ const struct sound_focus_param sound_focus_data,
+ const struct source_tracking_param source_tracking_data)
+{
+ int i = 0, len = 0;
+ char value[MAX_STR_SIZE] = "";
+
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES) {
+ for (i = 0; i < MAX_SECTORS; i++) {
+ if ((i >=4) && (sound_focus_data.start_angle[i] == 0xFFFF))
+ continue;
+ if (i)
+ snprintf(value + strlen(value), MAX_STR_SIZE, ",");
+ snprintf(value + strlen(value), MAX_STR_SIZE, "%d", sound_focus_data.start_angle[i]);
+ }
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES, value);
+ }
+
+ strlcpy(value, "", sizeof(""));
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS) {
+ for (i = 0; i < MAX_SECTORS; i++) {
+ if ((i >=4) && (sound_focus_data.enable[i] == 0xFF))
+ continue;
+ if (i)
+ snprintf(value + strlen(value), MAX_STR_SIZE, ",");
+ snprintf(value + strlen(value), MAX_STR_SIZE, "%d", sound_focus_data.enable[i]);
+ }
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS, value);
+ }
+
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP)
+ str_parms_add_int(reply, AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP, sound_focus_data.gain_step);
+
+ strlcpy(value, "", sizeof(""));
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD) {
+ for (i = 0; i < MAX_SECTORS; i++) {
+ if ((i >=4) && (source_tracking_data.vad[i] == 0xFF))
+ continue;
+ if (i)
+ snprintf(value + strlen(value), MAX_STR_SIZE, ",");
+ snprintf(value + strlen(value), MAX_STR_SIZE, "%d", source_tracking_data.vad[i]);
+ }
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_SOURCE_TRACK_VAD, value);
+ }
+
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH)
+ str_parms_add_int(reply, AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_SPEECH, source_tracking_data.doa_speech);
+
+ strlcpy(value, "", sizeof(""));
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE) {
+ snprintf(value, MAX_STR_SIZE,
+ "%d,%d,%d", source_tracking_data.doa_noise[0], source_tracking_data.doa_noise[1], source_tracking_data.doa_noise[2]);
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_SOURCE_TRACK_DOA_NOISE, value);
+ }
+
+ strlcpy(value, "", sizeof(""));
+ if (bitmask & BITMASK_AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY) {
+ for (i = 0; i < 360; i++) {
+ if (i)
+ snprintf(value + strlen(value), MAX_STR_SIZE, ",");
+ snprintf(value + strlen(value), MAX_STR_SIZE, "%d", source_tracking_data.polar_activity[i]);
+ }
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_SOURCE_TRACK_POLAR_ACTIVITY, value);
+ }
+}
+
+void audio_extn_source_track_get_parameters(struct audio_device *adev,
+ struct str_parms *query,
+ struct str_parms *reply)
+{
+ int bitmask = 0, ret = 0;
+ struct sound_focus_param sound_focus_data;
+ struct source_tracking_param source_tracking_data;
+
+ memset(&sound_focus_data, 0xFF, sizeof(struct sound_focus_param));
+ memset(&source_tracking_data, 0xFF, sizeof(struct source_tracking_param));
+
+ // Parse the input parameters string for Source Tracking keys
+ bitmask = parse_soundfocus_sourcetracking_keys(query);
+ if (bitmask) {
+ // Get the parameter values from the backend
+ ret = get_soundfocus_sourcetracking_data(adev, bitmask, &sound_focus_data, &source_tracking_data);
+ if (ret == 0) {
+ // Construct the return string with key, value pairs
+ send_soundfocus_sourcetracking_params(reply, bitmask, sound_focus_data, source_tracking_data);
+ }
+ }
+}
+
+void audio_extn_source_track_set_parameters(struct audio_device *adev,
+ struct str_parms *parms)
+{
+ int len, ret, count;;
+ struct mixer_ctl *ctl;
+ char sound_focus_mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = "Sound Focus";
+ char *value = NULL;
+ char *kv_pairs = str_parms_to_str(parms);
+
+ len = strlen(kv_pairs);
+ value = (char*)calloc(len, sizeof(char));
+ if(value == NULL) {
+ ret = -ENOMEM;
+ ALOGE("%s: failed to allocate memory", __func__);
+
+ goto done;
+ }
+
+ // Parse the input parameter string for Source Tracking key, value pairs
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES,
+ value, len);
+ if (ret >= 0) {
+ char *saveptr, *tok;
+ int i = 0, val;
+ struct sound_focus_param sound_focus_param;
+
+ memset(&sound_focus_param, 0xFF, sizeof(struct sound_focus_param));
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_START_ANGLES);
+ tok = strtok_r(value, ",", &saveptr);
+ while ((i < MAX_SECTORS) && (tok != NULL)) {
+ if (sscanf(tok, "%d", &val) == 1) {
+ sound_focus_param.start_angle[i++] = (uint16_t)val;
+ }
+ tok = strtok_r(NULL, ",", &saveptr);
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS,
+ value, len);
+ if (ret >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_ENABLE_SECTORS);
+ tok = strtok_r(value, ",", &saveptr);
+ i = 0;
+ while ((i < MAX_SECTORS) && (tok != NULL)) {
+ if (sscanf(tok, "%d", &val) == 1) {
+ sound_focus_param.enable[i++] = (uint8_t)val;
+ }
+ tok = strtok_r(NULL, ",", &saveptr);
+ }
+ } else {
+ ALOGE("%s: SoundFocus.enable_sectors key not found", __func__);
+
+ goto done;
+ }
+
+ ret = str_parms_get_int(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP, &val);
+ if (ret >= 0) {
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SOUND_FOCUS_GAIN_STEP);
+ sound_focus_param.gain_step = (uint16_t)val;
+ } else {
+ ALOGE("%s: SoundFocus.gain_step key not found", __func__);
+
+ goto done;
+ }
+
+ /* Derive the mixer control name based on the use case and the audio h/w
+ * interface name for the corresponding audio device
+ */
+ ret = derive_mixer_ctl_from_usecase_intf(adev, sound_focus_mixer_ctl_name);
+ if (ret != 0) {
+ ALOGE("%s: Could not set Sound Focus Params", __func__);
+
+ goto done;
+ } else {
+ ALOGV("%s: Mixer Ctl name: %s", __func__, sound_focus_mixer_ctl_name);
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, sound_focus_mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, sound_focus_mixer_ctl_name);
+
+ goto done;
+ } else {
+ ALOGV("%s: Setting Sound Focus Params", __func__);
+
+ for (i = 0; i < MAX_SECTORS;i++) {
+ ALOGV("%s: start_angles[%d] = %d", __func__, i, sound_focus_param.start_angle[i]);
+ }
+ for (i = 0; i < MAX_SECTORS;i++) {
+ ALOGV("%s: enable_sectors[%d] = %d", __func__, i, sound_focus_param.enable[i]);
+ }
+ ALOGV("%s: gain_step = %d", __func__, sound_focus_param.gain_step);
+
+ mixer_ctl_update(ctl);
+ count = mixer_ctl_get_num_values(ctl);
+ if (count != sizeof(struct sound_focus_param)) {
+ ALOGE("%s: mixer_ctl_get_num_values() invalid data size", __func__);
+
+ goto done;
+ }
+
+ // Set the parameters on the mixer control derived above
+ ret = mixer_ctl_set_array(ctl, (void *)&sound_focus_param, count);
+ if (ret != 0) {
+ ALOGE("%s: mixer_ctl_set_array() failed to set Sound Focus Params", __func__);
+
+ goto done;
+ }
+ }
+ }
+
+done:
+ free(kv_pairs);
+ if(value != NULL)
+ free(value);
+ return;
+}
+#endif /* SOURCE_TRACKING_ENABLED end */
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index c2d3400..138bd3c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -190,6 +190,8 @@
[USECASE_VOLTE_CALL] = "volte-call",
[USECASE_QCHAT_CALL] = "qchat-call",
[USECASE_VOWLAN_CALL] = "vowlan-call",
+ [USECASE_VOICEMMODE1_CALL] = "voicemmode1-call",
+ [USECASE_VOICEMMODE2_CALL] = "voicemmode2-call",
[USECASE_COMPRESS_VOIP_CALL] = "compress-voip-call",
[USECASE_INCALL_REC_UPLINK] = "incall-rec-uplink",
[USECASE_INCALL_REC_DOWNLINK] = "incall-rec-downlink",
@@ -285,7 +287,9 @@
format == AUDIO_FORMAT_AAC_HE_V2 ||
format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
- format == AUDIO_FORMAT_FLAC)
+ format == AUDIO_FORMAT_FLAC ||
+ format == AUDIO_FORMAT_WMA ||
+ format == AUDIO_FORMAT_WMA_PRO)
return true;
return false;
@@ -308,6 +312,12 @@
case AUDIO_FORMAT_FLAC:
id = SND_AUDIOCODEC_FLAC;
break;
+ case AUDIO_FORMAT_WMA:
+ id = SND_AUDIOCODEC_WMA;
+ break;
+ case AUDIO_FORMAT_WMA_PRO:
+ id = SND_AUDIOCODEC_WMA_PRO;
+ break;
default:
ALOGE("%s: Unsupported audio format :%x", __func__, format);
}
@@ -660,6 +670,7 @@
if (usecase->type != PCM_PLAYBACK &&
usecase != uc_info &&
usecase->in_snd_device != snd_device &&
+ (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
@@ -739,14 +750,15 @@
return ret;
}
-static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
+audio_usecase_t get_usecase_id_from_usecase_type(struct audio_device *adev,
+ usecase_type_t type)
{
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == VOICE_CALL) {
+ if (usecase->type == type) {
ALOGV("%s: usecase id %d", __func__, usecase->id);
return usecase->id;
}
@@ -803,7 +815,7 @@
*/
if (voice_is_in_call(adev) && adev->mode == AUDIO_MODE_IN_CALL) {
vc_usecase = get_usecase_from_list(adev,
- get_voice_usecase_id_from_list(adev));
+ get_usecase_id_from_usecase_type(adev, VOICE_CALL));
if ((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
(usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
in_snd_device = vc_usecase->in_snd_device;
@@ -1796,6 +1808,52 @@
}
}
+ if (out->format == AUDIO_FORMAT_WMA || out->format == AUDIO_FORMAT_WMA_PRO) {
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_FORMAT_TAG, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->format = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_BLOCK_ALIGN, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->options.wma.super_block_align = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_BIT_PER_SAMPLE, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->options.wma.bits_per_sample = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_CHANNEL_MASK, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->options.wma.channelmask = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->options.wma.encodeopt = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION1, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->options.wma.encodeopt1 = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION2, value, sizeof(value));
+ if (ret >= 0) {
+ out->compr_config.codec->options.wma.encodeopt2 = atoi(value);
+ out->send_new_metadata = 1;
+ }
+ ALOGV("WMA params: fmt %x, balgn %x, sr %d, chmsk %x, encop %x, op1 %x, op2 %x",
+ out->compr_config.codec->format,
+ out->compr_config.codec->options.wma.super_block_align,
+ out->compr_config.codec->options.wma.bits_per_sample,
+ out->compr_config.codec->options.wma.channelmask,
+ out->compr_config.codec->options.wma.encodeopt,
+ out->compr_config.codec->options.wma.encodeopt1,
+ out->compr_config.codec->options.wma.encodeopt2);
+ }
+
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
if(ret >= 0)
is_meta_data_params = true;
@@ -2131,6 +2189,7 @@
}
if (!out->playback_started && ret >= 0) {
compress_start(out->compr);
+ audio_extn_dts_eagle_fade(adev, true, out);
out->playback_started = 1;
out->offload_state = OFFLOAD_STATE_PLAYING;
@@ -3363,7 +3422,7 @@
struct audio_stream_in **stream_in,
audio_input_flags_t flags __unused,
const char *address __unused,
- audio_source_t source __unused)
+ audio_source_t source)
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_in *in;
@@ -3383,8 +3442,8 @@
}
ALOGD("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x)\
- stream_handle(%p) io_handle(%d)",__func__, config->sample_rate, config->channel_mask,
- devices, &in->stream, handle);
+ stream_handle(%p) io_handle(%d) source(%d)",__func__, config->sample_rate, config->channel_mask,
+ devices, &in->stream, handle, source);
pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
@@ -3405,7 +3464,7 @@
in->stream.get_input_frames_lost = in_get_input_frames_lost;
in->device = devices;
- in->source = AUDIO_SOURCE_DEFAULT;
+ in->source = source;
in->dev = adev;
in->standby = 1;
in->channel_mask = config->channel_mask;
@@ -3471,6 +3530,13 @@
channel_count,
is_low_latency);
in->config.period_size = buffer_size / frame_size;
+ if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+ (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ (voice_extn_compress_voip_is_format_supported(in->format)) &&
+ (in->config.rate == 8000 || in->config.rate == 16000) &&
+ (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
+ voice_extn_compress_voip_open_input_stream(in);
+ }
}
/* This stream could be for sound trigger lab,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index d0e01e5..67f5279 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -115,6 +115,8 @@
USECASE_VOLTE_CALL,
USECASE_QCHAT_CALL,
USECASE_VOWLAN_CALL,
+ USECASE_VOICEMMODE1_CALL,
+ USECASE_VOICEMMODE2_CALL,
USECASE_COMPRESS_VOIP_CALL,
USECASE_INCALL_REC_UPLINK,
@@ -342,6 +344,8 @@
int pcm_ioctl(struct pcm *pcm, int request, ...);
int get_snd_card_state(struct audio_device *adev);
+audio_usecase_t get_usecase_id_from_usecase_type(struct audio_device *adev,
+ usecase_type_t type);
#define LITERAL_TO_STRING(x) #x
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index 689e834..69c9341 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -216,8 +216,32 @@
hw_info->snd_devices = NULL;
hw_info->num_snd_devices = 0;
strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8952-snd-card")) {
+ strlcpy(hw_info->type, "", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8952", sizeof(hw_info->name));
+ hw_info->snd_devices = NULL;
+ hw_info->num_snd_devices = 0;
+ strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8952-snd-card-mtp")) {
+ strlcpy(hw_info->type, "", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8952", sizeof(hw_info->name));
+ hw_info->snd_devices = NULL;
+ hw_info->num_snd_devices = 0;
+ strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8952-tomtom-snd-card")) {
+ strlcpy(hw_info->type, "", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8952", sizeof(hw_info->name));
+ hw_info->snd_devices = NULL;
+ hw_info->num_snd_devices = 0;
+ strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8952-skum-snd-card")) {
+ strlcpy(hw_info->type, "", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8952", sizeof(hw_info->name));
+ hw_info->snd_devices = NULL;
+ hw_info->num_snd_devices = 0;
+ strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
} else {
- ALOGW("%s: Not an 8x16/8939/8909 device", __func__);
+ ALOGW("%s: Not an 8x16/8939/8909/8952 device", __func__);
}
}
@@ -232,7 +256,7 @@
}
if (strstr(snd_card_name, "msm8x16") || strstr(snd_card_name, "msm8939") ||
- strstr(snd_card_name, "msm8909")) {
+ strstr(snd_card_name, "msm8909") || strstr(snd_card_name, "msm8952")) {
ALOGV("8x16 - variant soundcard");
update_hardware_info_8x16(hw_info, snd_card_name);
} else {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index deec058..043615d 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -36,6 +36,7 @@
#include "edid.h"
#include "sound/compress_params.h"
#include "sound/msmcal-hwdep.h"
+#include <dirent.h>
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
@@ -49,6 +50,7 @@
#define MIXER_XML_PATH_SKUC "/system/etc/mixer_paths_skuc.xml"
#define MIXER_XML_PATH_SKUE "/system/etc/mixer_paths_skue.xml"
#define MIXER_XML_PATH_SKUL "/system/etc/mixer_paths_skul.xml"
+#define MIXER_XML_PATH_SKUM "/system/etc/mixer_paths_qrd_skum.xml"
#define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
#define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
#define MIXER_XML_PATH_I2S "/system/etc/mixer_paths_i2s.xml"
@@ -187,6 +189,7 @@
bool slowtalk;
bool hd_voice;
bool ec_ref_enabled;
+ bool is_wsa_speaker;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -207,6 +210,14 @@
bool edid_valid;
};
+static bool is_external_codec = false;
+static const int pcm_device_table_of_ext_codec[AUDIO_USECASE_MAX][2] = {
+ [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC, QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC}
+};
+
+/* List of use cases that has different PCM device ID's for internal and external codecs */
+static const int misc_usecase[AUDIO_USECASE_MAX] = { USECASE_QCHAT_CALL };
+
int pcm_device_table[AUDIO_USECASE_MAX][2] = {
[USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {DEEP_BUFFER_PCM_DEVICE,
DEEP_BUFFER_PCM_DEVICE},
@@ -281,6 +292,7 @@
[SND_DEVICE_OUT_SPEAKER] = "speaker",
[SND_DEVICE_OUT_SPEAKER_EXTERNAL_1] = "speaker-ext-1",
[SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = "speaker-ext-2",
+ [SND_DEVICE_OUT_SPEAKER_WSA] = "wsa-speaker",
[SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
[SND_DEVICE_OUT_HEADPHONES] = "headphones",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
@@ -288,6 +300,7 @@
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = "speaker-and-headphones-ext-2",
[SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
[SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = "wsa-voice-speaker",
[SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
[SND_DEVICE_OUT_HDMI] = "hdmi",
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
@@ -383,6 +396,7 @@
[SND_DEVICE_OUT_SPEAKER] = 14,
[SND_DEVICE_OUT_SPEAKER_EXTERNAL_1] = 14,
[SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = 14,
+ [SND_DEVICE_OUT_SPEAKER_WSA] = 135,
[SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
[SND_DEVICE_OUT_HEADPHONES] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
@@ -390,6 +404,7 @@
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = 10,
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = 135,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
[SND_DEVICE_OUT_HDMI] = 18,
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
@@ -487,6 +502,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_2)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
@@ -494,6 +510,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_WSA)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
@@ -668,6 +685,32 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+static bool is_misc_usecase(audio_usecase_t usecase) {
+ bool ret = false;
+ int i;
+
+ for (i = 0; i < AUDIO_USECASE_MAX; i++) {
+ if(usecase == misc_usecase[i]) {
+ ret = true;
+ break;
+ }
+ }
+ return ret;
+}
+
+
+static void update_codec_type(const char *snd_card_name) {
+
+ if (!strncmp(snd_card_name, "msm8939-tapan-snd-card",
+ sizeof("msm8939-tapan-snd-card")) ||
+ !strncmp(snd_card_name, "msm8939-tapan9302-snd-card",
+ sizeof("msm8939-tapan9302-snd-card"))||
+ !strncmp(snd_card_name, "msm8939-tomtom9330-snd-card",
+ sizeof("msm8939-tomtom9330-snd-card"))) {
+ ALOGI("%s: snd_card_name: %s",__func__,snd_card_name);
+ is_external_codec = true;
+ }
+}
static void query_platform(const char *snd_card_name,
char *mixer_xml_path)
{
@@ -789,6 +832,28 @@
msm_device_to_be_id = msm_device_to_be_id_internal_codec;
msm_be_id_array_len =
sizeof(msm_device_to_be_id_external_codec) / sizeof(msm_device_to_be_id_internal_codec[0]);
+ } else if (!strncmp(snd_card_name, "msm8952-snd-card-mtp",
+ sizeof("msm8952-snd-card-mtpmsm8952-snd-card-mtp"))) {
+ strlcpy(mixer_xml_path, MIXER_XML_PATH_MTP,
+ sizeof(MIXER_XML_PATH_MTP));
+ msm_device_to_be_id = msm_device_to_be_id_internal_codec;
+ msm_be_id_array_len =
+ sizeof(msm_device_to_be_id_internal_codec) / sizeof(msm_device_to_be_id_internal_codec[0]);
+ } else if (!strncmp(snd_card_name, "msm8952-tomtom-snd-card",
+ sizeof("msm8952-tomtom-snd-card"))) {
+ strlcpy(mixer_xml_path, MIXER_XML_PATH_WCD9330,
+ sizeof(MIXER_XML_PATH_WCD9330));
+ 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, "msm8952-skum-snd-card",
+ sizeof("msm8952-skum-snd-card"))) {
+ strlcpy(mixer_xml_path, MIXER_XML_PATH_SKUM,
+ sizeof(MIXER_XML_PATH_SKUM));
+ msm_device_to_be_id = msm_device_to_be_id_internal_codec;
+ msm_be_id_array_len =
+ sizeof(msm_device_to_be_id_internal_codec) / sizeof(msm_device_to_be_id_internal_codec[0]);
+
} else {
strlcpy(mixer_xml_path, MIXER_XML_PATH,
sizeof(MIXER_XML_PATH));
@@ -1170,6 +1235,7 @@
return NULL;
}
adev->snd_card = snd_card_num;
+ update_codec_type(snd_card_name);
ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
break;
}
@@ -1196,6 +1262,7 @@
my_data->slowtalk = false;
my_data->hd_voice = false;
my_data->edid_info = NULL;
+ my_data->is_wsa_speaker = false;
property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1235,6 +1302,7 @@
property_get("persist.audio.FFSP.enable", ffspEnable, "");
if (!strncmp("true", ffspEnable, sizeof("true"))) {
acdb_device_table[SND_DEVICE_OUT_SPEAKER] = 131;
+ acdb_device_table[SND_DEVICE_OUT_SPEAKER_WSA] = 131;
acdb_device_table[SND_DEVICE_OUT_SPEAKER_REVERSE] = 131;
acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 131;
acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131;
@@ -1309,6 +1377,25 @@
}
audio_extn_pm_vote();
+ // Check if WSA speaker is supported in codec
+ char CodecPeek[1024] = "/sys/kernel/debug/asoc/";
+ DIR *dir;
+ struct dirent *dirent;
+ char file_name[10] = "wsa";
+ strcat(CodecPeek, snd_card_name);
+
+ dir = opendir(CodecPeek);
+ if (dir != NULL) {
+ while (NULL != (dirent = readdir(dir))) {
+ if (strstr (dirent->d_name,file_name))
+ {
+ my_data->is_wsa_speaker = true;
+ break;
+ }
+ }
+ closedir(dir);
+ }
+
acdb_init_fail:
set_platform_defaults();
@@ -1413,11 +1500,19 @@
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type)
{
- int device_id;
- if (device_type == PCM_PLAYBACK)
- device_id = pcm_device_table[usecase][0];
- else
- device_id = pcm_device_table[usecase][1];
+ int device_id = -1;
+
+ if (is_external_codec && is_misc_usecase(usecase)) {
+ if (device_type == PCM_PLAYBACK)
+ device_id = pcm_device_table_of_ext_codec[usecase][0];
+ else
+ device_id = pcm_device_table_of_ext_codec[usecase][1];
+ } else {
+ if (device_type == PCM_PLAYBACK)
+ device_id = pcm_device_table[usecase][0];
+ else
+ device_id = pcm_device_table[usecase][1];
+ }
return device_id;
}
@@ -1966,7 +2061,10 @@
else
snd_device = SND_DEVICE_OUT_BT_SCO;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ if (my_data->is_wsa_speaker)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -2019,7 +2117,12 @@
if (adev->speaker_lr_swap)
snd_device = SND_DEVICE_OUT_SPEAKER_REVERSE;
else
- snd_device = SND_DEVICE_OUT_SPEAKER;
+ {
+ if (my_data->is_wsa_speaker)
+ snd_device = SND_DEVICE_OUT_SPEAKER_WSA;
+ else
+ snd_device = SND_DEVICE_OUT_SPEAKER;
+ }
}
} else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
if (adev->bt_wb_speech_enabled)
@@ -3187,8 +3290,15 @@
ALOGV("%s bit width: %d, sample rate: %d", __func__, bit_width, sample_rate);
int ret = 0;
+ const char *snd_card_name = mixer_get_name(adev->mixer);
if (bit_width != adev->cur_codec_backend_bit_width) {
- const char * mixer_ctl_name = "SLIM_0_RX Format";
+ const char * mixer_ctl_name;
+ if (!strncmp(snd_card_name, "msm8952-tomtom-snd-card",
+ sizeof("msm8952-tomtom-snd-card"))) {
+ mixer_ctl_name = "SLIM_0_RX Format";
+ }
+ else
+ mixer_ctl_name = "MI2S_RX Format";
struct mixer_ctl *ctl;
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -3891,3 +4001,56 @@
strlcpy(buf, PLATFORM_IMAGE_NAME, sizeof(PLATFORM_IMAGE_NAME));
return 0;
}
+
+/*
+ * This is a lookup table to map android audio input device to audio h/w interface (backend).
+ * The table can be extended for other input devices by adding appropriate entries.
+ * Also the audio interface for a particular input device can be overriden by adding
+ * corresponding entry in audio_platform_info.xml file.
+ */
+struct audio_device_to_audio_interface audio_device_to_interface_table[] = {
+ {AUDIO_DEVICE_IN_BUILTIN_MIC, ENUM_TO_STRING(AUDIO_DEVICE_IN_BUILTIN_MIC), "TERT_MI2S"},
+ {AUDIO_DEVICE_IN_BACK_MIC, ENUM_TO_STRING(AUDIO_DEVICE_IN_BACK_MIC), "TERT_MI2S"},
+};
+
+int audio_device_to_interface_table_len =
+ sizeof(audio_device_to_interface_table) / sizeof(audio_device_to_interface_table[0]);
+
+
+int platform_set_audio_device_interface(const char * device_name,
+ const char *intf_name)
+{
+ int ret = 0;
+ int i;
+
+ if (device_name == NULL || intf_name == NULL) {
+ ALOGE("%s: Invalid input", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ALOGD("%s: Enter, device name:%s, intf name:%s", __func__, device_name, intf_name);
+
+ size_t device_name_len = strlen(device_name);
+ for (i = 0; i < audio_device_to_interface_table_len; i++) {
+ char* name = audio_device_to_interface_table[i].device_name;
+ size_t name_len = strlen(name);
+ if ((name_len == device_name_len) &&
+ (strncmp(device_name, name, name_len) == 0)) {
+ ALOGD("%s: Matched device name:%s, overwrite intf name with %s",
+ __func__, device_name, intf_name);
+
+ strlcpy(audio_device_to_interface_table[i].interface_name, intf_name,
+ sizeof(audio_device_to_interface_table[i].interface_name));
+ goto done;
+ }
+ }
+ ALOGE("%s: Could not find matching device name %s",
+ __func__, device_name);
+
+ ret = -EINVAL;
+
+done:
+ return ret;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index dfb976d..03a02bf 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -58,12 +58,14 @@
SND_DEVICE_OUT_SPEAKER_EXTERNAL_1,
SND_DEVICE_OUT_SPEAKER_EXTERNAL_2,
SND_DEVICE_OUT_SPEAKER_REVERSE,
+ SND_DEVICE_OUT_SPEAKER_WSA,
SND_DEVICE_OUT_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2,
SND_DEVICE_OUT_VOICE_HANDSET,
SND_DEVICE_OUT_VOICE_SPEAKER,
+ SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_HDMI,
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
@@ -219,17 +221,6 @@
#define PLAYBACK_OFFLOAD_DEVICE 9
#ifdef MULTIPLE_OFFLOAD_ENABLED
-#ifdef PLATFORM_APQ8084
-#define PLAYBACK_OFFLOAD_DEVICE2 17
-#define PLAYBACK_OFFLOAD_DEVICE3 18
-#define PLAYBACK_OFFLOAD_DEVICE4 34
-#define PLAYBACK_OFFLOAD_DEVICE5 35
-#define PLAYBACK_OFFLOAD_DEVICE6 36
-#define PLAYBACK_OFFLOAD_DEVICE7 37
-#define PLAYBACK_OFFLOAD_DEVICE8 38
-#define PLAYBACK_OFFLOAD_DEVICE9 39
-#endif
-#if defined (PLATFORM_MSM8994)
#define PLAYBACK_OFFLOAD_DEVICE2 17
#define PLAYBACK_OFFLOAD_DEVICE3 18
#define PLAYBACK_OFFLOAD_DEVICE4 37
@@ -239,7 +230,6 @@
#define PLAYBACK_OFFLOAD_DEVICE8 41
#define PLAYBACK_OFFLOAD_DEVICE9 42
#endif
-#endif
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
@@ -253,7 +243,8 @@
#define VOICE_CALL_PCM_DEVICE 2
#define VOICE2_CALL_PCM_DEVICE 13
#define VOLTE_CALL_PCM_DEVICE 15
-#define QCHAT_CALL_PCM_DEVICE 14
+#define QCHAT_CALL_PCM_DEVICE 26
+#define QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC 28
#define VOWLAN_CALL_PCM_DEVICE 16
#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
@@ -310,4 +301,12 @@
*/
#define DEVICE_PARAM_MUTE_ID 0
#define DEVICE_PARAM_LATENCY_ID 1
+
+#define ENUM_TO_STRING(X) #X
+
+struct audio_device_to_audio_interface {
+ audio_devices_t device;
+ char device_name[100];
+ char interface_name[100];
+};
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 4c798b6..bbfa042 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -1153,3 +1153,9 @@
{
return 0;
}
+
+int platform_set_audio_device_interface(const char * device_name __unused,
+ const char *intf_name __unused)
+{
+ return -ENOSYS;
+}
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index d34a733..24e23e6 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -44,9 +44,13 @@
#include "sound/msmcal-hwdep.h"
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
-#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
+#define MIXER_XML_DEFAULT_PATH "/system/etc/mixer_paths.xml"
#define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
#define MIXER_XML_PATH_I2S "/system/etc/mixer_paths_i2s.xml"
+#define MIXER_XML_BASE_STRING "/system/etc/mixer_paths"
+#define MIXER_FILE_DELIMITER "_"
+#define MIXER_FILE_EXT ".xml"
+
#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/system/etc/audio_platform_info_i2s.xml"
@@ -242,6 +246,10 @@
[USECASE_VOLTE_CALL] = {VOLTE_CALL_PCM_DEVICE, VOLTE_CALL_PCM_DEVICE},
[USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE, QCHAT_CALL_PCM_DEVICE},
[USECASE_VOWLAN_CALL] = {VOWLAN_CALL_PCM_DEVICE, VOWLAN_CALL_PCM_DEVICE},
+ [USECASE_VOICEMMODE1_CALL] = {VOICEMMODE1_CALL_PCM_DEVICE,
+ VOICEMMODE1_CALL_PCM_DEVICE},
+ [USECASE_VOICEMMODE2_CALL] = {VOICEMMODE2_CALL_PCM_DEVICE,
+ VOICEMMODE2_CALL_PCM_DEVICE},
[USECASE_COMPRESS_VOIP_CALL] = {COMPRESS_VOIP_CALL_PCM_DEVICE, COMPRESS_VOIP_CALL_PCM_DEVICE},
[USECASE_INCALL_REC_UPLINK] = {AUDIO_RECORD_PCM_DEVICE,
AUDIO_RECORD_PCM_DEVICE},
@@ -637,6 +645,32 @@
{AUDIO_DEVICE_NONE , -1},
{AUDIO_DEVICE_OUT_DEFAULT , -1},
};
+#elif PLATFORM_MSM8996
+static int msm_device_to_be_id [][NO_COLS] = {
+ {AUDIO_DEVICE_OUT_EARPIECE , 2},
+ {AUDIO_DEVICE_OUT_SPEAKER , 2},
+ {AUDIO_DEVICE_OUT_WIRED_HEADSET , 2},
+ {AUDIO_DEVICE_OUT_WIRED_HEADPHONE , 2},
+ {AUDIO_DEVICE_OUT_BLUETOOTH_SCO , 11},
+ {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET , 11},
+ {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT , 11},
+ {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP , -1},
+ {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES , -1},
+ {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER , -1},
+ {AUDIO_DEVICE_OUT_AUX_DIGITAL , 4},
+ {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET , 9},
+ {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET , 9},
+ {AUDIO_DEVICE_OUT_USB_ACCESSORY , -1},
+ {AUDIO_DEVICE_OUT_USB_DEVICE , -1},
+ {AUDIO_DEVICE_OUT_REMOTE_SUBMIX , 9},
+ {AUDIO_DEVICE_OUT_PROXY , 9},
+/* Add the correct be ids */
+ {AUDIO_DEVICE_OUT_FM , 7},
+ {AUDIO_DEVICE_OUT_FM_TX , 8},
+ {AUDIO_DEVICE_OUT_ALL , -1},
+ {AUDIO_DEVICE_NONE , -1},
+ {AUDIO_DEVICE_OUT_DEFAULT , -1},
+};
#else
static int msm_device_to_be_id [][NO_COLS] = {
{AUDIO_DEVICE_NONE, -1},
@@ -988,9 +1022,12 @@
char baseband[PROPERTY_VALUE_MAX];
char value[PROPERTY_VALUE_MAX];
struct platform_data *my_data = NULL;
- int retry_num = 0, snd_card_num = 0, key = 0;
+ int retry_num = 0, snd_card_num = 0, key = 0, ret = 0;
const char *snd_card_name;
char *cvd_version = NULL;
+ char *snd_internal_name = NULL;
+ char *tmp = NULL;
+ char mixer_xml_file[MIXER_PATH_MAX_LENGTH]= {0};
my_data = calloc(1, sizeof(struct platform_data));
@@ -1028,10 +1065,51 @@
adev->audio_route = audio_route_init(snd_card_num,
MIXER_XML_PATH_I2S);
- } else if (audio_extn_read_xml(adev, snd_card_num, MIXER_XML_PATH,
- MIXER_XML_PATH_AUXPCM) == -ENOSYS) {
- adev->audio_route = audio_route_init(snd_card_num,
- MIXER_XML_PATH);
+ } else {
+ /* Get the codec internal name from the sound card name
+ * and form the mixer paths file name dynamically. This
+ * is generic way of picking any codec name based mixer
+ * files in future with no code change. This code
+ * assumes mixer files are formed with format as
+ * mixer_paths_internalcodecname.xml
+
+ * If this dynamically read mixer files fails to open then it
+ * falls back to default mixer file i.e mixer_paths.xml. This is
+ * done to preserve backward compatibility but not mandatory as
+ * long as the mixer files are named as per above assumption.
+ */
+
+ snd_internal_name = strtok_r(snd_card_name, "-", &tmp);
+ if (snd_internal_name != NULL)
+ snd_internal_name = strtok_r(NULL, "-", &tmp);
+
+ if (snd_internal_name != NULL) {
+ strlcpy(mixer_xml_file, MIXER_XML_BASE_STRING,
+ MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_xml_file, MIXER_FILE_DELIMITER,
+ MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_xml_file, snd_internal_name,
+ MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_xml_file, MIXER_FILE_EXT,
+ MIXER_PATH_MAX_LENGTH);
+ } else {
+ strlcpy(mixer_xml_file, MIXER_XML_DEFAULT_PATH,
+ MIXER_PATH_MAX_LENGTH);
+ }
+
+ if (F_OK == access(mixer_xml_file, 0)) {
+ ALOGD("%s: Loading mixer file: %s", __func__, mixer_xml_file);
+ if (audio_extn_read_xml(adev, snd_card_num, mixer_xml_file,
+ MIXER_XML_PATH_AUXPCM) == -ENOSYS)
+ adev->audio_route = audio_route_init(snd_card_num,
+ mixer_xml_file);
+ } else {
+ ALOGD("%s: Loading default mixer file", __func__);
+ if(audio_extn_read_xml(adev, snd_card_num, MIXER_XML_DEFAULT_PATH,
+ MIXER_XML_PATH_AUXPCM) == -ENOSYS)
+ adev->audio_route = audio_route_init(snd_card_num,
+ MIXER_XML_DEFAULT_PATH);
+ }
}
if (!adev->audio_route) {
ALOGE("%s: Failed to init audio route controls, aborting.",
@@ -3366,10 +3444,15 @@
ALOGE("%s: Invalid channel mapping used", __func__);
return -EINVAL;
}
- strlcpy(mixer_ctl_name, "Playback Channel Map", sizeof(mixer_ctl_name));
+
+ /*
+ * If snd_id is greater than 0, stream channel mapping
+ * If snd_id is below 0, typically -1, device channel mapping
+ */
if (snd_id >= 0) {
- snprintf(device_num, sizeof(device_num), "%d", snd_id);
- strncat(mixer_ctl_name, device_num, 13);
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "Playback Channel Map%d", snd_id);
+ } else {
+ strlcpy(mixer_ctl_name, "Playback Device Channel Map", sizeof(mixer_ctl_name));
}
ALOGD("%s mixer_ctl_name:%s", __func__, mixer_ctl_name);
@@ -3659,3 +3742,54 @@
end:
return ret;
}
+
+/*
+ * This is a lookup table to map android audio input device to audio h/w interface (backend).
+ * The table can be extended for other input devices by adding appropriate entries.
+ * Also the audio interface for a particular input device can be overriden by adding
+ * corresponding entry in audio_platform_info.xml file.
+ */
+struct audio_device_to_audio_interface audio_device_to_interface_table[] = {
+ {AUDIO_DEVICE_IN_BUILTIN_MIC, ENUM_TO_STRING(AUDIO_DEVICE_IN_BUILTIN_MIC), "SLIMBUS_0"},
+ {AUDIO_DEVICE_IN_BACK_MIC, ENUM_TO_STRING(AUDIO_DEVICE_IN_BACK_MIC), "SLIMBUS_0"},
+};
+
+int audio_device_to_interface_table_len =
+ sizeof(audio_device_to_interface_table) / sizeof(audio_device_to_interface_table[0]);
+
+int platform_set_audio_device_interface(const char *device_name, const char *intf_name)
+{
+ int ret = 0;
+ int i;
+
+ if (device_name == NULL || intf_name == NULL) {
+ ALOGE("%s: Invalid input", __func__);
+
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ALOGD("%s: Enter, device name:%s, intf name:%s", __func__, device_name, intf_name);
+
+ size_t device_name_len = strlen(device_name);
+ for (i = 0; i < audio_device_to_interface_table_len; i++) {
+ char* name = audio_device_to_interface_table[i].device_name;
+ size_t name_len = strlen(name);
+ if ((name_len == device_name_len) &&
+ (strncmp(device_name, name, name_len) == 0)) {
+ ALOGD("%s: Matched device name:%s, overwrite intf name with %s",
+ __func__, device_name, intf_name);
+
+ strlcpy(audio_device_to_interface_table[i].interface_name, intf_name,
+ sizeof(audio_device_to_interface_table[i].interface_name));
+ goto done;
+ }
+ }
+ ALOGE("%s: Could not find matching device name %s",
+ __func__, device_name);
+
+ ret = -EINVAL;
+
+done:
+ return ret;
+}
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 7deb6da..83922d5 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -280,6 +280,12 @@
#define VOLTE_CALL_PCM_DEVICE 14
#define QCHAT_CALL_PCM_DEVICE 20
#define VOWLAN_CALL_PCM_DEVICE 36
+#elif PLATFORM_MSM8996
+#define VOICE_CALL_PCM_DEVICE 40
+#define VOICE2_CALL_PCM_DEVICE 41
+#define VOLTE_CALL_PCM_DEVICE 14
+#define QCHAT_CALL_PCM_DEVICE 20
+#define VOWLAN_CALL_PCM_DEVICE 33
#else
#define VOICE_CALL_PCM_DEVICE 2
#define VOICE2_CALL_PCM_DEVICE 22
@@ -288,6 +294,14 @@
#define VOWLAN_CALL_PCM_DEVICE 36
#endif
+#ifdef PLATFORM_MSM8996
+#define VOICEMMODE1_CALL_PCM_DEVICE 2
+#define VOICEMMODE2_CALL_PCM_DEVICE 22
+#else
+#define VOICEMMODE1_CALL_PCM_DEVICE 44
+#define VOICEMMODE2_CALL_PCM_DEVICE 45
+#endif
+
#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
#define AFE_PROXY_RECORD_PCM_DEVICE 8
@@ -360,4 +374,12 @@
*/
#define DEVICE_PARAM_MUTE_ID 0
#define DEVICE_PARAM_LATENCY_ID 1
+
+#define ENUM_TO_STRING(X) #X
+
+struct audio_device_to_audio_interface {
+ audio_devices_t device;
+ char device_name[100];
+ char interface_name[100];
+};
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 3e81d68..3808b14 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -109,4 +109,5 @@
void platform_invalidate_edid(void * platform);
int platform_set_hdmi_config(struct stream_out *out);
int platform_set_device_params(struct stream_out *out, int param, int value);
+int platform_set_audio_device_interface(const char * device_name, const char *intf_name);
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/platform_info.c b/hal/platform_info.c
index a176177..13a314e 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-2015, 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
@@ -46,6 +46,7 @@
BITWIDTH,
PCM_ID,
BACKEND_NAME,
+ INTERFACE_NAME,
} section_t;
typedef void (* section_process_fn)(const XML_Char **attr);
@@ -54,6 +55,7 @@
static void process_bit_width(const XML_Char **attr);
static void process_pcm_id(const XML_Char **attr);
static void process_backend_name(const XML_Char **attr);
+static void process_interface_name(const XML_Char **attr);
static void process_root(const XML_Char **attr);
static section_process_fn section_table[] = {
@@ -62,6 +64,7 @@
[BITWIDTH] = process_bit_width,
[PCM_ID] = process_pcm_id,
[BACKEND_NAME] = process_backend_name,
+ [INTERFACE_NAME] = process_interface_name,
};
static section_t section;
@@ -83,6 +86,11 @@
* ...
* ...
* </pcm_ids>
+ * <interface_names>
+ * <device name="Use audio device name here, not sound device name" interface="PRIMARY_I2S"/>
+ * ...
+ * ...
+ * </interface_names>
* </audio_platform_info>
*/
@@ -237,6 +245,34 @@
return;
}
+static void process_interface_name(const XML_Char **attr)
+{
+ int ret;
+
+ if (strcmp(attr[0], "name") != 0) {
+ ALOGE("%s: 'name' not found, no Audio Interface set!", __func__);
+
+ goto done;
+ }
+
+ if (strcmp(attr[2], "interface") != 0) {
+ ALOGE("%s: Device %s has no Audio Interface set!",
+ __func__, attr[1]);
+
+ goto done;
+ }
+
+ ret = platform_set_audio_device_interface((char *)attr[1], (char *)attr[3]);
+ if (ret < 0) {
+ ALOGE("%s: Audio Interface not set!", __func__);
+
+ goto done;
+ }
+
+done:
+ return;
+}
+
static void start_tag(void *userdata __unused, const XML_Char *tag_name,
const XML_Char **attr)
{
@@ -252,9 +288,12 @@
section = PCM_ID;
} else if (strcmp(tag_name, "backend_names") == 0) {
section = BACKEND_NAME;
+ } else if (strcmp(tag_name, "interface_names") == 0) {
+ section = INTERFACE_NAME;
} else if (strcmp(tag_name, "device") == 0) {
- if ((section != ACDB) && (section != BACKEND_NAME) && (section != BITWIDTH)) {
- ALOGE("device tag only supported for acdb/backend names");
+ if ((section != ACDB) && (section != BACKEND_NAME) && (section != BITWIDTH) &&
+ (section != INTERFACE_NAME)) {
+ ALOGE("device tag only supported for acdb/backend names/bitwitdh/interface names");
return;
}
@@ -284,6 +323,8 @@
section = ROOT;
} else if (strcmp(tag_name, "backend_names") == 0) {
section = ROOT;
+ } else if (strcmp(tag_name, "interface_names") == 0) {
+ section = ROOT;
}
}
diff --git a/hal/voice.c b/hal/voice.c
index 5bc0a4e..9fc1081 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -157,6 +157,8 @@
}
ALOGD("voice_config.rate %d\n", voice_config.rate);
+ voice_set_mic_mute(adev, adev->voice.mic_mute);
+
ALOGV("%s: Opening PCM playback device card_id(%d) device_id(%d)",
__func__, adev->snd_card, pcm_dev_rx_id);
session->pcm_rx = pcm_open(adev->snd_card,
@@ -293,6 +295,18 @@
session_id, rec_mode);
ALOGV("%s: Update usecase to %d",__func__, in->usecase);
} else {
+ /*
+ * Reject the recording instances, where the recording is started
+ * with In-call voice recording source types but voice call is not
+ * active by the time input is started
+ */
+ if ((in->source == AUDIO_SOURCE_VOICE_UPLINK) ||
+ (in->source == AUDIO_SOURCE_VOICE_DOWNLINK) ||
+ (in->source == AUDIO_SOURCE_VOICE_CALL)) {
+ ret = -EINVAL;
+ ALOGE("%s: As voice call is not active, Incall rec usecase can't be \
+ selected for requested source:%d",__func__, in->source);
+ }
ALOGV("%s: voice call not active", __func__);
}
diff --git a/hal/voice.h b/hal/voice.h
index 5038dfa..5a9cce1 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -24,7 +24,7 @@
#define VOICE_SESS_IDX (BASE_SESS_IDX)
#ifdef MULTI_VOICE_SESSION_ENABLED
-#define MAX_VOICE_SESSIONS 5
+#define MAX_VOICE_SESSIONS 7
#else
#define MAX_VOICE_SESSIONS 1
#endif
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index e8ba1f9..14af6fc 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -697,6 +697,10 @@
voip_data.sample_rate = in->config.rate;
}
+ ret = voip_set_mode(in->dev, in->format);
+ if (ret < 0)
+ goto done;
+
in->usecase = USECASE_COMPRESS_VOIP_CALL;
if (in->config.rate == 16000)
in->config = pcm_config_voip_wb;
@@ -704,7 +708,6 @@
in->config = pcm_config_voip_nb;
voip_data.in_stream_count++;
- ret = voip_set_mode(in->dev, in->format);
done:
ALOGV("%s: exit, ret=%d", __func__, ret);
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 00068fe..b806bab 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -44,17 +44,21 @@
#define VOICE_EXTN_PARAMETER_VALUE_MAX_LEN 256
-#define VOICE2_VSID 0x10DC1000
-#define VOLTE_VSID 0x10C02000
-#define QCHAT_VSID 0x10803000
-#define VOWLAN_VSID 0x10002000
-#define ALL_VSID 0xFFFFFFFF
+#define VOICE2_VSID 0x10DC1000
+#define VOLTE_VSID 0x10C02000
+#define QCHAT_VSID 0x10803000
+#define VOWLAN_VSID 0x10002000
+#define VOICEMMODE1_VSID 0x11C05000
+#define VOICEMMODE2_VSID 0x11DC5000
+#define ALL_VSID 0xFFFFFFFF
/* Voice Session Indices */
#define VOICE2_SESS_IDX (VOICE_SESS_IDX + 1)
#define VOLTE_SESS_IDX (VOICE_SESS_IDX + 2)
#define QCHAT_SESS_IDX (VOICE_SESS_IDX + 3)
#define VOWLAN_SESS_IDX (VOICE_SESS_IDX + 4)
+#define MMODE1_SESS_IDX (VOICE_SESS_IDX + 5)
+#define MMODE2_SESS_IDX (VOICE_SESS_IDX + 6)
/* Call States */
#define CALL_HOLD (BASE_CALL_STATE + 2)
@@ -87,6 +91,8 @@
vsid == VOICE2_VSID ||
vsid == VOLTE_VSID ||
vsid == QCHAT_VSID ||
+ vsid == VOICEMMODE1_VSID ||
+ vsid == VOICEMMODE2_VSID ||
vsid == VOWLAN_VSID)
return true;
else
@@ -118,6 +124,14 @@
usecase_id = USECASE_VOWLAN_CALL;
break;
+ case MMODE1_SESS_IDX:
+ usecase_id = USECASE_VOICEMMODE1_CALL;
+ break;
+
+ case MMODE2_SESS_IDX:
+ usecase_id = USECASE_VOICEMMODE2_CALL;
+ break;
+
default:
ALOGE("%s: Invalid voice session index\n", __func__);
}
@@ -346,6 +360,8 @@
adev->voice.session[VOLTE_SESS_IDX].vsid = VOLTE_VSID;
adev->voice.session[QCHAT_SESS_IDX].vsid = QCHAT_VSID;
adev->voice.session[VOWLAN_SESS_IDX].vsid = VOWLAN_VSID;
+ adev->voice.session[MMODE1_SESS_IDX].vsid = VOICEMMODE1_VSID;
+ adev->voice.session[MMODE2_SESS_IDX].vsid = VOICEMMODE2_VSID;
}
int voice_extn_get_session_from_use_case(struct audio_device *adev,
@@ -375,6 +391,14 @@
*session = &adev->voice.session[VOWLAN_SESS_IDX];
break;
+ case USECASE_VOICEMMODE1_CALL:
+ *session = &adev->voice.session[MMODE1_SESS_IDX];
+ break;
+
+ case USECASE_VOICEMMODE2_CALL:
+ *session = &adev->voice.session[MMODE2_SESS_IDX];
+ break;
+
default:
ALOGE("%s: Invalid usecase_id:%d\n", __func__, usecase_id);
*session = NULL;
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 880838a..be70166 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -26,7 +26,8 @@
LOCAL_SHARED_LIBRARIES := \
libcutils \
liblog \
- libtinyalsa
+ libtinyalsa \
+ libdl
LOCAL_MODULE_TAGS := optional
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index 57cf8ef..ad1e7c9 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -17,14 +17,17 @@
* limitations under the License.
*/
-#define LOG_TAG "offload_effect_bass_boost"
+#define LOG_TAG "offload_effect_bass"
//#define LOG_NDEBUG 0
#include <cutils/list.h>
#include <cutils/log.h>
+#include <cutils/properties.h>
#include <tinyalsa/asoundlib.h>
#include <sound/audio_effects.h>
#include <audio_effects/effect_bassboost.h>
+#include <stdlib.h>
+#include <dlfcn.h>
#include "effect_api.h"
#include "bass_boost.h"
@@ -41,6 +44,243 @@
"The Android Open Source Project",
};
+#define LIB_ACDB_LOADER "libacdbloader.so"
+#define PBE_CONF_APP_ID 0x00011134
+
+enum {
+ AUDIO_DEVICE_CAL_TYPE = 0,
+ AUDIO_STREAM_CAL_TYPE,
+};
+
+typedef struct acdb_audio_cal_cfg {
+ uint32_t persist;
+ uint32_t snd_dev_id;
+ uint32_t dev_id;
+ int32_t acdb_dev_id;
+ uint32_t app_type;
+ uint32_t topo_id;
+ uint32_t sampling_rate;
+ uint32_t cal_type;
+ uint32_t module_id;
+ uint32_t param_id;
+} acdb_audio_cal_cfg_t;
+
+typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
+static int pbe_load_config(struct pbe_params *params);
+
+/*
+ * Bass operations
+ */
+int bass_get_parameter(effect_context_t *context, effect_param_t *p,
+ uint32_t *size)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+ int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
+ int32_t *param_tmp = (int32_t *)p->data;
+ int32_t param = *param_tmp++;
+ void *value = p->data + voffset;
+ int i;
+
+ ALOGV("%s", __func__);
+
+ p->status = 0;
+
+ switch (param) {
+ case BASSBOOST_PARAM_STRENGTH_SUPPORTED:
+ if (p->vsize < sizeof(uint32_t))
+ p->status = -EINVAL;
+ p->vsize = sizeof(uint32_t);
+ break;
+ case BASSBOOST_PARAM_STRENGTH:
+ if (p->vsize < sizeof(int16_t))
+ p->status = -EINVAL;
+ p->vsize = sizeof(int16_t);
+ break;
+ default:
+ p->status = -EINVAL;
+ }
+
+ *size = sizeof(effect_param_t) + voffset + p->vsize;
+
+ if (p->status != 0)
+ return 0;
+
+ switch (param) {
+ case BASSBOOST_PARAM_STRENGTH_SUPPORTED:
+ ALOGV("%s: BASSBOOST_PARAM_STRENGTH_SUPPORTED", __func__);
+ if (bass_ctxt->active_index == BASS_BOOST)
+ *(uint32_t *)value = 1;
+ else
+ *(uint32_t *)value = 0;
+ break;
+
+ case BASSBOOST_PARAM_STRENGTH:
+ ALOGV("%s: BASSBOOST_PARAM_STRENGTH", __func__);
+ if (bass_ctxt->active_index == BASS_BOOST)
+ *(int16_t *)value = bassboost_get_strength(&(bass_ctxt->bassboost_ctxt));
+ else
+ *(int16_t *)value = 0;
+ break;
+
+ default:
+ p->status = -EINVAL;
+ break;
+ }
+
+ return 0;
+}
+
+int bass_set_parameter(effect_context_t *context, effect_param_t *p,
+ uint32_t size __unused)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+ int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
+ void *value = p->data + voffset;
+ int32_t *param_tmp = (int32_t *)p->data;
+ int32_t param = *param_tmp++;
+ uint32_t strength;
+
+ ALOGV("%s", __func__);
+
+ p->status = 0;
+
+ switch (param) {
+ case BASSBOOST_PARAM_STRENGTH:
+ ALOGV("%s BASSBOOST_PARAM_STRENGTH", __func__);
+ if (bass_ctxt->active_index == BASS_BOOST) {
+ strength = (uint32_t)(*(int16_t *)value);
+ bassboost_set_strength(&(bass_ctxt->bassboost_ctxt), strength);
+ } else {
+ /* stength supported only for BB and not for PBE, but do not
+ * return error for unsupported case, as it fails cts test
+ */
+ ALOGD("%s ignore set strength, index %d",
+ __func__, bass_ctxt->active_index);
+ break;
+ }
+ break;
+ default:
+ p->status = -EINVAL;
+ break;
+ }
+
+ return 0;
+}
+
+int bass_set_device(effect_context_t *context, uint32_t device)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ if (device == AUDIO_DEVICE_OUT_SPEAKER) {
+ bass_ctxt->active_index = BASS_PBE;
+ ALOGV("%s: set PBE mode, device: %x", __func__, device);
+ } else if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ device == AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES) {
+ ALOGV("%s: set BB mode, device: %x", __func__, device);
+ bass_ctxt->active_index = BASS_BOOST;
+ } else {
+ ALOGI("%s: disabled by device: %x", __func__, device);
+ bass_ctxt->active_index = BASS_INVALID;
+ }
+
+ bassboost_set_device((effect_context_t *)&(bass_ctxt->bassboost_ctxt), device);
+ pbe_set_device((effect_context_t *)&(bass_ctxt->pbe_ctxt), device);
+
+ return 0;
+}
+
+int bass_reset(effect_context_t *context)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ bassboost_reset((effect_context_t *)&(bass_ctxt->bassboost_ctxt));
+ pbe_reset((effect_context_t *)&(bass_ctxt->pbe_ctxt));
+
+ return 0;
+}
+
+int bass_init(effect_context_t *context)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ // convery i/o channel config to sub effects
+ bass_ctxt->bassboost_ctxt.common.config = context->config;
+ bass_ctxt->pbe_ctxt.common.config = context->config;
+
+ ALOGV("%s", __func__);
+
+ bass_ctxt->active_index = BASS_BOOST;
+
+
+ bassboost_init((effect_context_t *)&(bass_ctxt->bassboost_ctxt));
+ pbe_init((effect_context_t *)&(bass_ctxt->pbe_ctxt));
+
+ return 0;
+}
+
+int bass_enable(effect_context_t *context)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ ALOGV("%s", __func__);
+
+ bassboost_enable((effect_context_t *)&(bass_ctxt->bassboost_ctxt));
+ pbe_enable((effect_context_t *)&(bass_ctxt->pbe_ctxt));
+
+ return 0;
+}
+
+int bass_disable(effect_context_t *context)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ ALOGV("%s", __func__);
+
+ bassboost_disable((effect_context_t *)&(bass_ctxt->bassboost_ctxt));
+ pbe_disable((effect_context_t *)&(bass_ctxt->pbe_ctxt));
+
+ return 0;
+}
+
+int bass_start(effect_context_t *context, output_context_t *output)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ ALOGV("%s", __func__);
+
+ bassboost_start((effect_context_t *)&(bass_ctxt->bassboost_ctxt), output);
+ pbe_start((effect_context_t *)&(bass_ctxt->pbe_ctxt), output);
+
+ return 0;
+}
+
+int bass_stop(effect_context_t *context, output_context_t *output)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ ALOGV("%s", __func__);
+
+ bassboost_stop((effect_context_t *)&(bass_ctxt->bassboost_ctxt), output);
+ pbe_stop((effect_context_t *)&(bass_ctxt->pbe_ctxt), output);
+
+ return 0;
+}
+
+int bass_set_mode(effect_context_t *context, int32_t hw_acc_fd)
+{
+ bass_context_t *bass_ctxt = (bass_context_t *)context;
+
+ ALOGV("%s", __func__);
+
+ bassboost_set_mode((effect_context_t *)&(bass_ctxt->bassboost_ctxt), hw_acc_fd);
+ pbe_set_mode((effect_context_t *)&(bass_ctxt->pbe_ctxt), hw_acc_fd);
+
+ return 0;
+}
+
+#undef LOG_TAG
+#define LOG_TAG "offload_effect_bb"
/*
* Bassboost operations
*/
@@ -70,99 +310,30 @@
return 0;
}
-int bassboost_get_parameter(effect_context_t *context, effect_param_t *p,
- uint32_t *size)
-{
- bassboost_context_t *bass_ctxt = (bassboost_context_t *)context;
- int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
- int32_t *param_tmp = (int32_t *)p->data;
- int32_t param = *param_tmp++;
- void *value = p->data + voffset;
- int i;
-
- ALOGV("%s: ctxt %p, param %d", __func__, bass_ctxt, param);
-
- p->status = 0;
-
- switch (param) {
- case BASSBOOST_PARAM_STRENGTH_SUPPORTED:
- if (p->vsize < sizeof(uint32_t))
- p->status = -EINVAL;
- p->vsize = sizeof(uint32_t);
- break;
- case BASSBOOST_PARAM_STRENGTH:
- if (p->vsize < sizeof(int16_t))
- p->status = -EINVAL;
- p->vsize = sizeof(int16_t);
- break;
- default:
- p->status = -EINVAL;
- }
-
- *size = sizeof(effect_param_t) + voffset + p->vsize;
-
- if (p->status != 0)
- return 0;
-
- switch (param) {
- case BASSBOOST_PARAM_STRENGTH_SUPPORTED:
- *(uint32_t *)value = 1;
- break;
-
- case BASSBOOST_PARAM_STRENGTH:
- *(int16_t *)value = bassboost_get_strength(bass_ctxt);
- break;
-
- default:
- p->status = -EINVAL;
- break;
- }
-
- return 0;
-}
-
-int bassboost_set_parameter(effect_context_t *context, effect_param_t *p,
- uint32_t size __unused)
-{
- bassboost_context_t *bass_ctxt = (bassboost_context_t *)context;
- int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
- void *value = p->data + voffset;
- int32_t *param_tmp = (int32_t *)p->data;
- int32_t param = *param_tmp++;
- uint32_t strength;
-
- ALOGV("%s: ctxt %p, param %d", __func__, bass_ctxt, param);
-
- p->status = 0;
-
- switch (param) {
- case BASSBOOST_PARAM_STRENGTH:
- strength = (uint32_t)(*(int16_t *)value);
- bassboost_set_strength(bass_ctxt, strength);
- break;
- default:
- p->status = -EINVAL;
- break;
- }
-
- return 0;
-}
-
int bassboost_set_device(effect_context_t *context, uint32_t device)
{
bassboost_context_t *bass_ctxt = (bassboost_context_t *)context;
ALOGV("%s: ctxt %p, device 0x%x", __func__, bass_ctxt, device);
bass_ctxt->device = device;
- if((device == AUDIO_DEVICE_OUT_SPEAKER) ||
- (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) ||
- (device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER) ||
-#ifdef AFE_PROXY_ENABLED
- (device == AUDIO_DEVICE_OUT_PROXY) ||
-#endif
- (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
- (device == AUDIO_DEVICE_OUT_USB_ACCESSORY) ||
- (device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET)) {
+ if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ device == AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES) {
+ if (bass_ctxt->temp_disabled) {
+ if (effect_is_active(&bass_ctxt->common)) {
+ offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), true);
+ if (bass_ctxt->ctl)
+ offload_bassboost_send_params(bass_ctxt->ctl,
+ &bass_ctxt->offload_bass,
+ OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG);
+ if (bass_ctxt->hw_acc_fd > 0)
+ hw_acc_bassboost_send_params(bass_ctxt->hw_acc_fd,
+ &bass_ctxt->offload_bass,
+ OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG);
+ }
+ bass_ctxt->temp_disabled = false;
+ }
+ } else {
if (!bass_ctxt->temp_disabled) {
if (effect_is_active(&bass_ctxt->common)) {
offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), false);
@@ -178,21 +349,6 @@
bass_ctxt->temp_disabled = true;
}
ALOGI("%s: ctxt %p, disabled based on device", __func__, bass_ctxt);
- } else {
- if (bass_ctxt->temp_disabled) {
- if (effect_is_active(&bass_ctxt->common)) {
- offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), true);
- if (bass_ctxt->ctl)
- offload_bassboost_send_params(bass_ctxt->ctl,
- &bass_ctxt->offload_bass,
- OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG);
- if (bass_ctxt->hw_acc_fd > 0)
- hw_acc_bassboost_send_params(bass_ctxt->hw_acc_fd,
- &bass_ctxt->offload_bass,
- OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG);
- }
- bass_ctxt->temp_disabled = false;
- }
}
offload_bassboost_set_device(&(bass_ctxt->offload_bass), device);
return 0;
@@ -329,3 +485,234 @@
OFFLOAD_SEND_BASSBOOST_STRENGTH);
return 0;
}
+
+#undef LOG_TAG
+#define LOG_TAG "offload_effect_pbe"
+/*
+ * PBE operations
+ */
+
+int pbe_set_device(effect_context_t *context, uint32_t device)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+ char propValue[PROPERTY_VALUE_MAX];
+ bool pbe_enabled_by_prop = false;
+
+ ALOGV("%s: device: %d", __func__, device);
+ pbe_ctxt->device = device;
+
+ if (property_get("audio.safx.pbe.enabled", propValue, NULL)) {
+ pbe_enabled_by_prop = atoi(propValue) ||
+ !strncmp("true", propValue, 4);
+ }
+
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && pbe_enabled_by_prop == true) {
+ if (pbe_ctxt->temp_disabled) {
+ if (effect_is_active(&pbe_ctxt->common)) {
+ offload_pbe_set_enable_flag(&(pbe_ctxt->offload_pbe), true);
+ if (pbe_ctxt->ctl)
+ offload_pbe_send_params(pbe_ctxt->ctl,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ if (pbe_ctxt->hw_acc_fd > 0)
+ hw_acc_pbe_send_params(pbe_ctxt->hw_acc_fd,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ }
+ pbe_ctxt->temp_disabled = false;
+ }
+ } else {
+ if (!pbe_ctxt->temp_disabled) {
+ if (effect_is_active(&pbe_ctxt->common)) {
+ offload_pbe_set_enable_flag(&(pbe_ctxt->offload_pbe), false);
+ if (pbe_ctxt->ctl)
+ offload_pbe_send_params(pbe_ctxt->ctl,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG);
+ if (pbe_ctxt->hw_acc_fd > 0)
+ hw_acc_pbe_send_params(pbe_ctxt->hw_acc_fd,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG);
+ }
+ pbe_ctxt->temp_disabled = true;
+ }
+ }
+ offload_pbe_set_device(&(pbe_ctxt->offload_pbe), device);
+ return 0;
+}
+
+int pbe_reset(effect_context_t *context)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ return 0;
+}
+
+int pbe_init(effect_context_t *context)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ ALOGV("%s", __func__);
+ context->config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+ context->config.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
+ context->config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ context->config.inputCfg.samplingRate = 44100;
+ context->config.inputCfg.bufferProvider.getBuffer = NULL;
+ context->config.inputCfg.bufferProvider.releaseBuffer = NULL;
+ context->config.inputCfg.bufferProvider.cookie = NULL;
+ context->config.inputCfg.mask = EFFECT_CONFIG_ALL;
+ context->config.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
+ context->config.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
+ context->config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ context->config.outputCfg.samplingRate = 44100;
+ context->config.outputCfg.bufferProvider.getBuffer = NULL;
+ context->config.outputCfg.bufferProvider.releaseBuffer = NULL;
+ context->config.outputCfg.bufferProvider.cookie = NULL;
+ context->config.outputCfg.mask = EFFECT_CONFIG_ALL;
+
+ set_config(context, &context->config);
+
+ pbe_ctxt->hw_acc_fd = -1;
+ pbe_ctxt->temp_disabled = false;
+ memset(&(pbe_ctxt->offload_pbe), 0, sizeof(struct pbe_params));
+ pbe_load_config(&(pbe_ctxt->offload_pbe));
+
+ return 0;
+}
+
+int pbe_enable(effect_context_t *context)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ ALOGV("%s", __func__);
+
+ if (!offload_pbe_get_enable_flag(&(pbe_ctxt->offload_pbe)) &&
+ !(pbe_ctxt->temp_disabled)) {
+ offload_pbe_set_enable_flag(&(pbe_ctxt->offload_pbe), true);
+ if (pbe_ctxt->ctl)
+ offload_pbe_send_params(pbe_ctxt->ctl,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ if (pbe_ctxt->hw_acc_fd > 0)
+ hw_acc_pbe_send_params(pbe_ctxt->hw_acc_fd,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ }
+ return 0;
+}
+
+int pbe_disable(effect_context_t *context)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ ALOGV("%s", __func__);
+ if (offload_pbe_get_enable_flag(&(pbe_ctxt->offload_pbe))) {
+ offload_pbe_set_enable_flag(&(pbe_ctxt->offload_pbe), false);
+ if (pbe_ctxt->ctl)
+ offload_pbe_send_params(pbe_ctxt->ctl,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG);
+ if (pbe_ctxt->hw_acc_fd > 0)
+ hw_acc_pbe_send_params(pbe_ctxt->hw_acc_fd,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG);
+ }
+ return 0;
+}
+
+int pbe_start(effect_context_t *context, output_context_t *output)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ ALOGV("%s", __func__);
+ pbe_ctxt->ctl = output->ctl;
+ ALOGV("output->ctl: %p", output->ctl);
+ if (offload_pbe_get_enable_flag(&(pbe_ctxt->offload_pbe))) {
+ if (pbe_ctxt->ctl)
+ offload_pbe_send_params(pbe_ctxt->ctl, &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ if (pbe_ctxt->hw_acc_fd > 0)
+ hw_acc_pbe_send_params(pbe_ctxt->hw_acc_fd,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ }
+ return 0;
+}
+
+int pbe_stop(effect_context_t *context, output_context_t *output __unused)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ ALOGV("%s", __func__);
+ pbe_ctxt->ctl = NULL;
+ return 0;
+}
+
+int pbe_set_mode(effect_context_t *context, int32_t hw_acc_fd)
+{
+ pbe_context_t *pbe_ctxt = (pbe_context_t *)context;
+
+ ALOGV("%s: ctxt %p", __func__, pbe_ctxt);
+ pbe_ctxt->hw_acc_fd = hw_acc_fd;
+ if ((pbe_ctxt->hw_acc_fd > 0) &&
+ (offload_pbe_get_enable_flag(&(pbe_ctxt->offload_pbe))))
+ hw_acc_pbe_send_params(pbe_ctxt->hw_acc_fd,
+ &pbe_ctxt->offload_pbe,
+ OFFLOAD_SEND_PBE_ENABLE_FLAG |
+ OFFLOAD_SEND_PBE_CONFIG);
+ return 0;
+}
+
+static int pbe_load_config(struct pbe_params *params)
+{
+ int ret = 0;
+ uint32_t len = 0;
+ uint32_t propValue = 0;
+ uint32_t pbe_app_type = PBE_CONF_APP_ID;
+ char propValueStr[PROPERTY_VALUE_MAX];
+ void *acdb_handle = NULL;
+ acdb_get_audio_cal_t acdb_get_audio_cal = NULL;
+ acdb_audio_cal_cfg_t cal_cfg = {0};
+
+ acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
+ if (acdb_handle == NULL) {
+ ALOGE("%s error opening library %s", __func__, LIB_ACDB_LOADER);
+ return -EFAULT;
+ }
+
+ acdb_get_audio_cal = (acdb_get_audio_cal_t)dlsym(acdb_handle,
+ "acdb_loader_get_audio_cal_v2");
+ if (acdb_get_audio_cal == NULL) {
+ dlclose(acdb_handle);
+ ALOGE("%s error resolving acdb func symbols", __func__);
+ return -EFAULT;
+ }
+ if (property_get("audio.safx.pbe.app.type", propValueStr, "0")) {
+ propValue = atoll(propValueStr);
+ if (propValue != 0) {
+ pbe_app_type = propValue;
+ }
+ }
+ ALOGD("%s pbe_app_type = 0x%.8x", __func__, pbe_app_type);
+
+ cal_cfg.persist = 1;
+ cal_cfg.cal_type = AUDIO_STREAM_CAL_TYPE;
+ cal_cfg.app_type = pbe_app_type;
+ cal_cfg.module_id = PBE_CONF_MODULE_ID;
+ cal_cfg.param_id = PBE_CONF_PARAM_ID;
+
+ len = sizeof(params->config);
+ ret = acdb_get_audio_cal((void *)&cal_cfg, (void*)&(params->config), &len);
+ ALOGD("%s ret = %d, len = %u", __func__, ret, len);
+ if (ret == 0)
+ params->cfg_len = len;
+
+ dlclose(acdb_handle);
+ return ret;
+}
diff --git a/post_proc/bass_boost.h b/post_proc/bass_boost.h
index 6f0e61b..8bf51d3 100644
--- a/post_proc/bass_boost.h
+++ b/post_proc/bass_boost.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -22,6 +22,13 @@
#include "bundle.h"
+enum {
+ BASS_INVALID = -1,
+ BASS_BOOST = 0, // index of bassboost
+ BASS_PBE, // index of PBE
+ BASS_COUNT // totol number of bass type
+};
+
extern const effect_descriptor_t bassboost_descriptor;
typedef struct bassboost_context_s {
@@ -37,11 +44,50 @@
struct bass_boost_params offload_bass;
} bassboost_context_t;
-int bassboost_get_parameter(effect_context_t *context, effect_param_t *p,
- uint32_t *size);
+typedef struct pbe_context_s {
+ effect_context_t common;
-int bassboost_set_parameter(effect_context_t *context, effect_param_t *p,
- uint32_t size);
+ // Offload vars
+ struct mixer_ctl *ctl;
+ int hw_acc_fd;
+ bool temp_disabled;
+ uint32_t device;
+ struct pbe_params offload_pbe;
+} pbe_context_t;
+
+typedef struct bass_context_s {
+ effect_context_t common;
+ bassboost_context_t bassboost_ctxt;
+ pbe_context_t pbe_ctxt;
+ int active_index;
+} bass_context_t;
+
+int bass_get_parameter(effect_context_t *context, effect_param_t *p,
+ uint32_t *size);
+
+int bass_set_parameter(effect_context_t *context, effect_param_t *p,
+ uint32_t size);
+
+int bass_set_device(effect_context_t *context, uint32_t device);
+
+int bass_set_mode(effect_context_t *context, int32_t hw_acc_fd);
+
+int bass_reset(effect_context_t *context);
+
+int bass_init(effect_context_t *context);
+
+int bass_enable(effect_context_t *context);
+
+int bass_disable(effect_context_t *context);
+
+int bass_start(effect_context_t *context, output_context_t *output);
+
+int bass_stop(effect_context_t *context, output_context_t *output);
+
+
+int bassboost_get_strength(bassboost_context_t *context);
+
+int bassboost_set_strength(bassboost_context_t *context, uint32_t strength);
int bassboost_set_device(effect_context_t *context, uint32_t device);
@@ -59,4 +105,20 @@
int bassboost_stop(effect_context_t *context, output_context_t *output);
+int pbe_set_device(effect_context_t *context, uint32_t device);
+
+int pbe_set_mode(effect_context_t *context, int32_t hw_acc_fd);
+
+int pbe_reset(effect_context_t *context);
+
+int pbe_init(effect_context_t *context);
+
+int pbe_enable(effect_context_t *context);
+
+int pbe_disable(effect_context_t *context);
+
+int pbe_start(effect_context_t *context, output_context_t *output);
+
+int pbe_stop(effect_context_t *context, output_context_t *output);
+
#endif /* OFFLOAD_EFFECT_BASS_BOOST_H_ */
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index 606c66b..e38a41c 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -520,25 +520,26 @@
eq_ctxt->ctl = NULL;
} else if (memcmp(uuid, &bassboost_descriptor.uuid,
sizeof(effect_uuid_t)) == 0) {
- bassboost_context_t *bass_ctxt = (bassboost_context_t *)
- calloc(1, sizeof(bassboost_context_t));
+ bass_context_t *bass_ctxt = (bass_context_t *)
+ calloc(1, sizeof(bass_context_t));
if (bass_ctxt == NULL) {
return -ENOMEM;
}
context = (effect_context_t *)bass_ctxt;
- context->ops.init = bassboost_init;
- context->ops.reset = bassboost_reset;
- context->ops.set_parameter = bassboost_set_parameter;
- context->ops.get_parameter = bassboost_get_parameter;
- context->ops.set_device = bassboost_set_device;
- context->ops.set_hw_acc_mode = bassboost_set_mode;
- context->ops.enable = bassboost_enable;
- context->ops.disable = bassboost_disable;
- context->ops.start = bassboost_start;
- context->ops.stop = bassboost_stop;
+ context->ops.init = bass_init;
+ context->ops.reset = bass_reset;
+ context->ops.set_parameter = bass_set_parameter;
+ context->ops.get_parameter = bass_get_parameter;
+ context->ops.set_device = bass_set_device;
+ context->ops.set_hw_acc_mode = bass_set_mode;
+ context->ops.enable = bass_enable;
+ context->ops.disable = bass_disable;
+ context->ops.start = bass_start;
+ context->ops.stop = bass_stop;
context->desc = &bassboost_descriptor;
- bass_ctxt->ctl = NULL;
+ bass_ctxt->bassboost_ctxt.ctl = NULL;
+ bass_ctxt->pbe_ctxt.ctl = NULL;
} else if (memcmp(uuid, &virtualizer_descriptor.uuid,
sizeof(effect_uuid_t)) == 0) {
virtualizer_context_t *virt_ctxt = (virtualizer_context_t *)
@@ -847,8 +848,6 @@
} break;
case EFFECT_CMD_SET_PARAM: {
if (pCmdData == NULL ||
- cmdSize > (int)(sizeof(effect_param_t) + sizeof(uint32_t) +
- sizeof(uint32_t)) ||
cmdSize < (int)(sizeof(effect_param_t) + sizeof(uint32_t) +
sizeof(uint16_t)) ||
pReplyData == NULL || *replySize != sizeof(int32_t)) {
diff --git a/post_proc/effect_api.c b/post_proc/effect_api.c
index 7c1968e..e15db17 100644
--- a/post_proc/effect_api.c
+++ b/post_proc/effect_api.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, 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
@@ -114,7 +114,7 @@
return -EINVAL;
} else {
*ctl = mixer_get_ctl_by_name(*mixer, mixer_string);
- if (!ctl) {
+ if (!(*ctl)) {
ALOGE("mixer_get_ctl_by_name failed");
mixer_close(*mixer);
*mixer = NULL;
@@ -235,6 +235,83 @@
bassboost, param_send_flags);
}
+void offload_pbe_set_device(struct pbe_params *pbe,
+ uint32_t device)
+{
+ ALOGV("%s: device=%d", __func__, device);
+ pbe->device = device;
+}
+
+void offload_pbe_set_enable_flag(struct pbe_params *pbe,
+ bool enable)
+{
+ ALOGV("%s: enable=%d", __func__, enable);
+ pbe->enable_flag = enable;
+}
+
+int offload_pbe_get_enable_flag(struct pbe_params *pbe)
+{
+ ALOGV("%s: enabled=%d", __func__, pbe->enable_flag);
+ return pbe->enable_flag;
+}
+
+static int pbe_send_params(eff_mode_t mode, void *ctl,
+ struct pbe_params *pbe,
+ unsigned param_send_flags)
+{
+ int param_values[128] = {0};
+ int i, *p_param_values = param_values, *cfg = NULL;
+
+ ALOGV("%s: enabled=%d", __func__, pbe->enable_flag);
+ *p_param_values++ = PBE_MODULE;
+ *p_param_values++ = pbe->device;
+ *p_param_values++ = 0; /* num of commands*/
+ if (param_send_flags & OFFLOAD_SEND_PBE_ENABLE_FLAG) {
+ *p_param_values++ = PBE_ENABLE;
+ *p_param_values++ = CONFIG_SET;
+ *p_param_values++ = 0; /* start offset if param size if greater than 128 */
+ *p_param_values++ = PBE_ENABLE_PARAM_LEN;
+ *p_param_values++ = pbe->enable_flag;
+ param_values[2] += 1;
+ }
+ if (param_send_flags & OFFLOAD_SEND_PBE_CONFIG) {
+ *p_param_values++ = PBE_CONFIG;
+ *p_param_values++ = CONFIG_SET;
+ *p_param_values++ = 0; /* start offset if param size if greater than 128 */
+ *p_param_values++ = pbe->cfg_len;
+ cfg = (int *)&pbe->config;
+ for (i = 0; i < (int)pbe->cfg_len ; i+= sizeof(*p_param_values))
+ *p_param_values++ = *cfg++;
+ param_values[2] += 1;
+ }
+
+ if ((mode == OFFLOAD) && param_values[2] && ctl) {
+ mixer_ctl_set_array((struct mixer_ctl *)ctl, param_values,
+ ARRAY_SIZE(param_values));
+ } else if ((mode == HW_ACCELERATOR) && param_values[2] &&
+ ctl && *(int *)ctl) {
+ if (ioctl(*(int *)ctl, AUDIO_EFFECTS_SET_PP_PARAMS, param_values) < 0)
+ ALOGE("%s: sending h/w acc effects params fail[%d]", __func__, errno);
+ }
+
+ return 0;
+}
+
+int offload_pbe_send_params(struct mixer_ctl *ctl,
+ struct pbe_params *pbe,
+ unsigned param_send_flags)
+{
+ return pbe_send_params(OFFLOAD, (void *)ctl, pbe,
+ param_send_flags);
+}
+
+int hw_acc_pbe_send_params(int fd, struct pbe_params *pbe,
+ unsigned param_send_flags)
+{
+ return pbe_send_params(HW_ACCELERATOR, (void *)&fd,
+ pbe, param_send_flags);
+}
+
void offload_virtualizer_set_device(struct virtualizer_params *virtualizer,
uint32_t device)
{
@@ -891,6 +968,11 @@
uint32_t i;
ALOGV("%s", __func__);
+ if (!ctl) {
+ ALOGE("%s: ctl is NULL, return invalid", __func__);
+ return -EINVAL;
+ }
+
if (param_send_flags & OFFLOAD_SEND_HPX_STATE_OFF) {
*p_param_values++ = DTS_EAGLE_MODULE_ENABLE;
*p_param_values++ = 0; /* hpx off*/
@@ -899,7 +981,7 @@
*p_param_values++ = 1; /* hpx on*/
}
- if ((mode == OFFLOAD) && ctl)
+ if (mode == OFFLOAD)
mixer_ctl_set_array(ctl, param_values, ARRAY_SIZE(param_values));
else {
if (ioctl(*(int *)ctl, AUDIO_EFFECTS_SET_PP_PARAMS, param_values) < 0)
diff --git a/post_proc/effect_api.h b/post_proc/effect_api.h
index e05924a..ce0503a 100644
--- a/post_proc/effect_api.h
+++ b/post_proc/effect_api.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2015, 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
@@ -39,6 +39,21 @@
struct mixer_ctl **ctl);
void offload_close_mixer(struct mixer **mixer);
+
+#define OFFLOAD_SEND_PBE_ENABLE_FLAG (1 << 0)
+#define OFFLOAD_SEND_PBE_CONFIG (OFFLOAD_SEND_PBE_ENABLE_FLAG << 1)
+void offload_pbe_set_device(struct pbe_params *pbe,
+ uint32_t device);
+void offload_pbe_set_enable_flag(struct pbe_params *pbe,
+ bool enable);
+int offload_pbe_get_enable_flag(struct pbe_params *pbe);
+
+int offload_pbe_send_params(struct mixer_ctl *ctl,
+ struct pbe_params *pbe,
+ unsigned param_send_flags);
+int hw_acc_pbe_send_params(int fd,
+ struct pbe_params *pbe,
+ unsigned param_send_flags);
#define OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG (1 << 0)
#define OFFLOAD_SEND_BASSBOOST_STRENGTH \
(OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG << 1)