Merge "hal: Add sound card update API"
diff --git a/audiod/Android.mk b/audiod/Android.mk
index c382c9d..c89b3cd 100644
--- a/audiod/Android.mk
+++ b/audiod/Android.mk
@@ -1,7 +1,10 @@
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
+
+ifneq (,$(findstring $(PLATFORM_VERSION), 5.0 5.1 5.1.1))
include external/stlport/libstlport.mk
+endif
LOCAL_SRC_FILES:= \
audiod_main.cpp \
@@ -13,8 +16,11 @@
libcutils \
libutils \
libbinder \
- libmedia \
- libstlport
+ libmedia
+
+ifneq (,$(findstring $(PLATFORM_VERSION), 5.0 5.1 5.1.1))
+LOCAL_SHARED_LIBRARIES += libstlport
+endif
LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
diff --git a/audiod/AudioDaemon.cpp b/audiod/AudioDaemon.cpp
index b4857c4..6f38218 100644
--- a/audiod/AudioDaemon.cpp
+++ b/audiod/AudioDaemon.cpp
@@ -125,6 +125,7 @@
struct dirent* in_file;
int fd;
String8 path;
+ String8 d_name;
if ((dp = opendir(events_dir)) == NULL) {
ALOGE("Cannot open switch directory to get list of audio events %s", events_dir);
@@ -148,8 +149,9 @@
if (fd == -1) {
ALOGE("Open %s failed : %s", path.string(), strerror(errno));
} else {
- mAudioEvents.push_back(std::make_pair(in_file->d_name, fd));
- mAudioEventsStatus.push_back(std::make_pair(in_file->d_name, 0));
+ d_name = in_file->d_name;
+ mAudioEvents.push_back(std::make_pair(d_name, fd));
+ mAudioEventsStatus.push_back(std::make_pair(d_name, 0));
ALOGD("event status mAudioEventsStatus= %s",
mAudioEventsStatus[0].first.string());
}
diff --git a/hal/Android.mk b/hal/Android.mk
index e0149f9..315aab2 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -53,6 +53,7 @@
LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_EDID)),true)
+ LOCAL_CFLAGS += -DHDMI_EDID
LOCAL_SRC_FILES += edid.c
endif
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 7bbe2f8..8c893c6 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -46,6 +46,9 @@
#define AUDIO_FORMAT_PCM_24_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_8_24_BIT)
#define AUDIO_OFFLOAD_CODEC_FORMAT "music_offload_codec_format"
#define audio_is_offload_pcm(format) (0)
+#define OFFLOAD_USE_SMALL_BUFFER false
+#else
+#define OFFLOAD_USE_SMALL_BUFFER (info->use_small_bufs)
#endif
#ifndef AFE_PROXY_ENABLED
@@ -221,6 +224,7 @@
#define audio_extn_sound_trigger_set_parameters(adev, parms) (0)
#define audio_extn_sound_trigger_check_and_get_session(in) (0)
#define audio_extn_sound_trigger_stop_lab(in) (0)
+#define audio_extn_sound_trigger_read(in, buffer, bytes) (0)
#else
enum st_event_type {
@@ -241,6 +245,8 @@
struct str_parms *parms);
void audio_extn_sound_trigger_check_and_get_session(struct stream_in *in);
void audio_extn_sound_trigger_stop_lab(struct stream_in *in);
+int audio_extn_sound_trigger_read(struct stream_in *in, void *buffer,
+ size_t bytes);
#endif
#ifndef AUXPCM_BT_ENABLED
@@ -316,6 +322,9 @@
void audio_extn_dolby_set_dmid(struct audio_device *adev);
#else
#define audio_extn_dolby_set_dmid(adev) (0)
+#define AUDIO_CHANNEL_OUT_PENTA (AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER)
+#define AUDIO_CHANNEL_OUT_SURROUND (AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT | \
+ AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_BACK_CENTER)
#endif
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 9051334..3c16c88 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -30,6 +30,7 @@
/* #define LOG_NDEBUG 0 */
#define LOG_NDDEBUG 0
+#include <errno.h>
#include <stdbool.h>
#include <stdlib.h>
#include <dlfcn.h>
@@ -64,7 +65,7 @@
{
struct sound_trigger_info *st_ses_info = NULL;
struct listnode *node;
- ALOGD("%s: list %d capture_handle %d", __func__,
+ ALOGV("%s: list empty %d capture_handle %d", __func__,
list_empty(&st_dev->st_ses_list), capture_handle);
list_for_each(node, &st_dev->st_ses_list) {
st_ses_info = node_to_item(node, struct sound_trigger_info , list);
@@ -128,6 +129,45 @@
return status;
}
+int audio_extn_sound_trigger_read(struct stream_in *in, void *buffer,
+ size_t bytes)
+{
+ int ret = -1;
+ struct sound_trigger_info *st_info = NULL;
+ audio_event_info_t event;
+
+ if (!st_dev)
+ return ret;
+
+ if (!in->is_st_session_active) {
+ ALOGE(" %s: Sound trigger is not active", __func__);
+ goto exit;
+ }
+ if(in->standby)
+ in->standby = false;
+
+ pthread_mutex_lock(&st_dev->lock);
+ st_info = get_sound_trigger_info(in->capture_handle);
+ pthread_mutex_unlock(&st_dev->lock);
+ if (st_info) {
+ event.u.aud_info.ses_info = &st_info->st_ses;
+ event.u.aud_info.buf = buffer;
+ event.u.aud_info.num_bytes = bytes;
+ ret = st_dev->st_callback(AUDIO_EVENT_READ_SAMPLES, &event);
+ }
+
+exit:
+ if (ret) {
+ if (-ENETRESET == ret)
+ in->is_st_session_active = false;
+ memset(buffer, 0, bytes);
+ ALOGV("%s: read failed status %d - sleep", __func__, ret);
+ usleep((bytes * 1000000) / (audio_stream_in_frame_size((struct audio_stream_in *)in) *
+ in->config.rate));
+ }
+ return ret;
+}
+
void audio_extn_sound_trigger_stop_lab(struct stream_in *in)
{
int status = 0;
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 5ea5a43..e703b13 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -485,6 +485,7 @@
enable_audio_route(adev, uc_info_tx);
pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
+ ALOGV("%s: pcm device id %d", __func__, pcm_dev_tx_id);
if (pcm_dev_tx_id < 0) {
ALOGE("%s: Invalid pcm device for usecase (%d)",
__func__, uc_info_tx->id);
@@ -647,6 +648,7 @@
char wsa_path[MAX_PATH] = {0};
int spk_1_tzn, spk_2_tzn;
char buf[32] = {0};
+ int ret;
/* If the value of this persist.spkr.cal.duration is 0
* then it means it will take 30min to calibrate
@@ -756,13 +758,17 @@
thermal_fd = open(wsa_path, O_RDONLY);
if (thermal_fd > 0) {
for (i = 0; i < NUM_ATTEMPTS; i++) {
- if (read(thermal_fd, buf, sizeof(buf))) {
+ if ((ret = read(thermal_fd, buf, sizeof(buf))) >= 0) {
t0_spk_1 = atoi(buf);
- if (i > 0 && (t0_spk_1 != t0_spk_prior))
+ if (i > 0 && (t0_spk_1 != t0_spk_prior)) {
+ ALOGE("%s: spkr1 curr temp: %d, prev temp: %d\n",
+ __func__, t0_spk_1, t0_spk_prior);
break;
+ }
t0_spk_prior = t0_spk_1;
+ sleep(1);
} else {
- ALOGE("%s: read fail for %s\n", __func__, wsa_path);
+ ALOGE("%s: read fail for %s err:%d\n", __func__, wsa_path, ret);
break;
}
}
@@ -773,26 +779,30 @@
if (i == NUM_ATTEMPTS) {
/*Convert temp into q6 format*/
t0_spk_1 = (t0_spk_1 * (1 << 6));
- ALOGE("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
+ ALOGD("%s: temp T0 for spkr1 %d\n", __func__, t0_spk_1);
} else {
- ALOGE("%s: thermal equilibrium failed for spkr1 in %d readings\n",
- __func__, NUM_ATTEMPTS);
+ ALOGD("%s: thermal equilibrium failed for spkr1 in %d/%d readings\n",
+ __func__, i, NUM_ATTEMPTS);
t0_spk_1 = SAFE_SPKR_TEMP_Q6;
}
}
if (spk_2_tzn >= 0) {
snprintf(wsa_path, MAX_PATH, TZ_WSA, spk_2_tzn);
- ALOGE("%s: wsa_path: %s\n", __func__, wsa_path);
+ ALOGD("%s: wsa_path: %s\n", __func__, wsa_path);
thermal_fd = open(wsa_path, O_RDONLY);
if (thermal_fd > 0) {
for (i = 0; i < NUM_ATTEMPTS; i++) {
- if (read(thermal_fd, buf, sizeof(buf))) {
+ if ((ret = read(thermal_fd, buf, sizeof(buf))) >= 0) {
t0_spk_2 = atoi(buf);
- if (i > 0 && (t0_spk_2 != t0_spk_prior))
+ if (i > 0 && (t0_spk_2 != t0_spk_prior)) {
+ ALOGE("%s: spkr2 curr temp: %d, prev temp: %d\n",
+ __func__, t0_spk_2, t0_spk_prior);
break;
+ }
t0_spk_prior = t0_spk_2;
+ sleep(1);
} else {
- ALOGE("%s: read fail for %s\n", __func__, wsa_path);
+ ALOGE("%s: read fail for %s err:%d\n", __func__, wsa_path, ret);
break;
}
}
@@ -803,10 +813,10 @@
if (i == NUM_ATTEMPTS) {
/*Convert temp into q6 format*/
t0_spk_2 = (t0_spk_2 * (1 << 6));
- ALOGE("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
+ ALOGD("%s: temp T0 for spkr2 %d\n", __func__, t0_spk_2);
} else {
- ALOGE("%s: thermal equilibrium failed for spkr2 in %d readings\n",
- __func__, NUM_ATTEMPTS);
+ ALOGE("%s: thermal equilibrium failed for spkr2 in %d/%d readings\n",
+ __func__, i, NUM_ATTEMPTS);
t0_spk_2 = SAFE_SPKR_TEMP_Q6;
}
}
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 3dbf159..7ad4811 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -71,7 +71,9 @@
#ifdef COMPRESS_VOIP_ENABLED
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_VOIP_RX),
#endif
+#ifdef HDMI_PASSTHROUGH_ENABLED
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
+#endif
};
const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -566,8 +568,11 @@
app_type_cfg[len++] = out->app_type_cfg.app_type;
app_type_cfg[len++] = acdb_dev_id;
if (((out->format == AUDIO_FORMAT_E_AC3) ||
- (out->format == AUDIO_FORMAT_E_AC3_JOC)) &&
- (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
+ (out->format == AUDIO_FORMAT_E_AC3_JOC))
+#ifdef HDMI_PASSTHROUGH_ENABLED
+ && (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)
+#endif
+ )
app_type_cfg[len++] = sample_rate * 4;
else
app_type_cfg[len++] = sample_rate;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9110fa8..4bc6b59 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -252,6 +252,26 @@
static int set_voice_volume_l(struct audio_device *adev, float volume);
+__attribute__ ((visibility ("default")))
+bool audio_hw_send_gain_dep_calibration(int level) {
+ bool ret_val = false;
+ ALOGV("%s: called ... ", __func__);
+
+ pthread_mutex_lock(&adev_init_lock);
+
+ if (adev != NULL && adev->platform != NULL) {
+ pthread_mutex_lock(&adev->lock);
+ ret_val = platform_send_gain_dep_cal(adev->platform, level);
+ pthread_mutex_unlock(&adev->lock);
+ } else {
+ ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform");
+ }
+
+ pthread_mutex_unlock(&adev_init_lock);
+
+ return ret_val;
+}
+
static int check_and_set_gapless_mode(struct audio_device *adev) {
@@ -623,7 +643,7 @@
ALOGV("%s: backend_idx: %d,"
"usecase_backend_idx: %d, curr device: %s, usecase device:"
"%s", __func__, backend_idx, usecase_backend_idx, platform_get_snd_device_name(snd_device),
- platform_get_snd_device_name(usecase->out_snd_device));
+ platform_get_snd_device_name(usecase->out_snd_device));
if (usecase->type != PCM_CAPTURE &&
(usecase->out_snd_device != snd_device || force_routing) &&
@@ -2185,7 +2205,7 @@
if (ret != 0) {
if (out->pcm)
- ALOGE("%s: error %ld - %s", __func__, ret, pcm_get_error(out->pcm));
+ ALOGE("%s: error %zu - %s", __func__, ret, pcm_get_error(out->pcm));
if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
voice_extn_compress_voip_close_output_stream(&out->stream.common);
@@ -2606,31 +2626,29 @@
pthread_mutex_lock(&in->lock);
- if (in->pcm) {
- if(SND_CARD_STATE_OFFLINE == snd_scard_state) {
- ALOGD(" %s: sound card is not active/SSR state", __func__);
- ret= -EIO;;
- goto exit;
- } else {
- if (in->is_st_session && !in->is_st_session_active) {
- ALOGD(" %s: Sound trigger is not active/SSR", __func__);
- ret= -EIO;;
- goto exit;
- }
- }
+ if (in->is_st_session) {
+ ALOGVV(" %s: reading on st session bytes=%zu", __func__, bytes);
+ /* Read from sound trigger HAL */
+ audio_extn_sound_trigger_read(in, buffer, bytes);
+ pthread_mutex_unlock(&in->lock);
+ return bytes;
+ }
+
+ if (in->pcm && (SND_CARD_STATE_OFFLINE == snd_scard_state)) {
+ ALOGD(" %s: sound card is not active/SSR state", __func__);
+ ret= -EIO;;
+ goto exit;
}
if (in->standby) {
- if (!in->is_st_session) {
- pthread_mutex_lock(&adev->lock);
- if (in->usecase == USECASE_COMPRESS_VOIP_CALL)
- ret = voice_extn_compress_voip_start_input_stream(in);
- else
- ret = start_input_stream(in);
- pthread_mutex_unlock(&adev->lock);
- if (ret != 0) {
- goto exit;
- }
+ pthread_mutex_lock(&adev->lock);
+ if (in->usecase == USECASE_COMPRESS_VOIP_CALL)
+ ret = voice_extn_compress_voip_start_input_stream(in);
+ else
+ ret = start_input_stream(in);
+ pthread_mutex_unlock(&adev->lock);
+ if (ret != 0) {
+ goto exit;
}
in->standby = 0;
}
@@ -2660,17 +2678,9 @@
exit:
/* ToDo: There may be a corner case when SSR happens back to back during
start/stop. Need to post different error to handle that. */
- if (-ENETRESET == ret) {
- /* CPE SSR results in kernel returning ENETRESET for sound trigger
- session reading on LAB data. In this case do not set sound card state
- offline, instead mark this sound trigger session inactive to avoid
- further reading of LAB data from CPE driver. Marking the session
- inactive handles both CPE and ADSP SSR for sound trigger session */
- if (!in->is_st_session)
- set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
- else
- in->is_st_session_active = false;
- }
+ if (-ENETRESET == ret)
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+
pthread_mutex_unlock(&in->lock);
if (ret != 0) {
@@ -2924,7 +2934,7 @@
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
- if (config->offload_info.use_small_bufs) {
+ if (platform_use_small_buffer(&config->offload_info)) {
//this flag is set from framework only if its for PCM formats
//no need to check for PCM format again
out->non_blocking = 0;
@@ -3385,7 +3395,7 @@
}
static int adev_open_input_stream(struct audio_hw_device *dev,
- audio_io_handle_t handle __unused,
+ audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 67f5279..5e61d36 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -38,6 +38,7 @@
#ifndef QCOM_AUDIO_HW_H
#define QCOM_AUDIO_HW_H
+#include <stdlib.h>
#include <cutils/list.h>
#include <hardware/audio.h>
#include <tinyalsa/asoundlib.h>
diff --git a/hal/edid.h b/hal/edid.h
index ec83ec8..0d7fbe6 100644
--- a/hal/edid.h
+++ b/hal/edid.h
@@ -92,5 +92,9 @@
int channel_allocation;
} edid_audio_info;
+#ifndef HDMI_EDID
+#define edid_get_sink_caps(info, edid_data) (0)
+#else
bool edid_get_sink_caps(edid_audio_info* info, char *edid_data);
+#endif
#endif /* EDID_H */
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 8f31b29..3a8f314 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -133,7 +133,6 @@
char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
[WCD9XXX_ANC_CAL] = "anc_cal",
[WCD9XXX_MBHC_CAL] = "mbhc_cal",
- [WCD9XXX_MAD_CAL] = "mad_cal",
};
#define AUDIO_PARAMETER_KEY_REC_PLAY_CONC "rec_play_conc_on"
@@ -175,6 +174,20 @@
typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
typedef int (*acdb_send_common_top_t) (void);
+typedef struct codec_backend_cfg {
+ uint32_t sample_rate;
+ uint32_t bit_width;
+ char *bitwidth_mixer_ctl;
+ char *samplerate_mixer_ctl;
+} codec_backend_cfg_t;
+
+typedef struct {
+ bool platform_na_prop_enabled;
+ bool ui_na_prop_enabled;
+} native_audio_prop;
+
+static native_audio_prop na_props = {0, 0};
+
struct platform_data {
struct audio_device *adev;
bool fluence_in_spkr_mode;
@@ -211,6 +224,7 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
+ codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
};
static bool is_external_codec = false;
@@ -294,6 +308,7 @@
[SND_DEVICE_OUT_SPEAKER_WSA] = "wsa-speaker",
[SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
[SND_DEVICE_OUT_HEADPHONES] = "headphones",
+ [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1] = "speaker-and-headphones-ext-1",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = "speaker-and-headphones-ext-2",
@@ -398,6 +413,7 @@
[SND_DEVICE_OUT_SPEAKER_WSA] = 135,
[SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
[SND_DEVICE_OUT_HEADPHONES] = 10,
+ [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = 10,
@@ -504,6 +520,7 @@
{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_HEADPHONES_44_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2)},
@@ -1036,7 +1053,7 @@
backend_table[dev] = NULL;
}
for (dev = 0; dev < SND_DEVICE_MAX; dev++) {
- backend_bit_width_table[dev] = 16;
+ backend_bit_width_table[dev] = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
}
// TBD - do these go to the platform-info.xml file.
@@ -1058,6 +1075,7 @@
backend_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
backend_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
backend_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
+ backend_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
}
void get_cvd_version(char *cvd_version, struct audio_device *adev)
@@ -1121,8 +1139,6 @@
struct wcdcal_ioctl_buffer codec_buffer;
struct param_data calib;
- if (!strcmp(cal_name_info[type], "mad_cal"))
- calib.acdb_id = SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID;
calib.get_size = 1;
ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
&calib);
@@ -1221,8 +1237,6 @@
const char *snd_card_name;
char mixer_xml_path[100],ffspEnable[PROPERTY_VALUE_MAX];
char *cvd_version = NULL;
-
- my_data = calloc(1, sizeof(struct platform_data));
if (!my_data) {
ALOGE("failed to allocate platform data");
return NULL;
@@ -1454,6 +1468,26 @@
audio_extn_dev_arbi_init();
my_data->edid_info = NULL;
+
+ /* initialize backend config */
+ for (idx = 0; idx < MAX_CODEC_BACKENDS; idx++) {
+ my_data->current_backend_cfg[idx].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ if (idx == HEADPHONE_44_1_BACKEND)
+ my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
+ my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ }
+
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_0_RX Format");
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_0_RX SampleRate");
+
+ my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_5_RX Format");
+ my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_5_RX SampleRate");
+
+ my_data->edid_info = NULL;
return my_data;
}
@@ -1731,14 +1765,107 @@
return backend_bit_width_table[snd_device];
}
-int platform_set_native_support(bool codec_support __unused)
+int platform_set_native_support(bool codec_support)
{
+ na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
+ = codec_support;
+ ALOGV("%s: na_props.platform_na_prop_enabled: %d", __func__,
+ na_props.platform_na_prop_enabled);
return 0;
}
-int platform_get_backend_index(snd_device_t snd_device __unused)
+int platform_get_native_support()
{
- return 0;
+ int ret;
+ if (na_props.platform_na_prop_enabled) {
+ ret = na_props.ui_na_prop_enabled;
+ } else {
+ ret = na_props.platform_na_prop_enabled;
+ }
+ ALOGV("%s: na_props.ui_na_prop_enabled: %d", __func__,
+ na_props.ui_na_prop_enabled);
+ return ret;
+}
+
+void native_audio_get_params(struct str_parms *query,
+ struct str_parms *reply,
+ char *value, int len)
+{
+ int ret;
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ value, len);
+ if (ret >= 0) {
+ if (na_props.platform_na_prop_enabled) {
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ na_props.ui_na_prop_enabled ? "true" : "false");
+ ALOGV("%s: na_props.ui_na_prop_enabled: %d", __func__,
+ na_props.ui_na_prop_enabled);
+ } else {
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ "false");
+ ALOGV("%s: native audio not supported: %d", __func__,
+ na_props.platform_na_prop_enabled);
+ }
+ }
+}
+
+int native_audio_set_params(struct platform_data *platform,
+ struct str_parms *parms, char *value, int len)
+{
+ int ret = 0;
+ struct audio_usecase *usecase;
+ struct listnode *node;
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ value, len);
+ if (ret >= 0) {
+ if (na_props.platform_na_prop_enabled) {
+ if (!strncmp("true", value, sizeof("true")))
+ na_props.ui_na_prop_enabled = true;
+ else
+ na_props.ui_na_prop_enabled = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO);
+
+ /* Iterate through the usecase list and trigger device switch for
+ all the appropriate usecases */
+ list_for_each(node, &(platform->adev)->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+
+ if (is_offload_usecase(usecase->id) &&
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) &&
+ OUTPUT_SAMPLING_RATE_44100 == usecase->stream.out->sample_rate) {
+ select_devices(platform->adev, usecase->id);
+ ALOGV("%s: triggering dynamic device switch for usecase: "
+ "%d, device: %d", __func__, usecase->id,
+ usecase->stream.out->devices);
+ }
+ }
+ } else {
+ ALOGV("%s: native audio not supported: %d", __func__,
+ na_props.platform_na_prop_enabled);
+ }
+ }
+ return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
+{
+ int32_t port = DEFAULT_CODEC_BACKEND;
+
+ if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ if (backend_table[snd_device] != NULL &&
+ !strcmp(backend_table[snd_device], "headphones-44.1"))
+ port = HEADPHONE_44_1_BACKEND;
+ else
+ port = DEFAULT_CODEC_BACKEND;
+ } else {
+ ALOGV("%s: Invalid device - %d ", __func__, snd_device);
+ }
+
+ ALOGV("%s: backend port - %d", __func__, port);
+ return port;
}
int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
@@ -2034,6 +2161,7 @@
audio_mode_t mode = adev->mode;
snd_device_t snd_device = SND_DEVICE_NONE;
audio_devices_t devices = out->devices;
+ unsigned int sample_rate = out->sample_rate;
#ifdef RECORD_PLAY_CONCURRENCY
bool use_voip_out_devices = false;
bool prop_rec_play_enabled = false;
@@ -2155,7 +2283,10 @@
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
- if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
+ if (OUTPUT_SAMPLING_RATE_44100 == sample_rate &&
+ platform_get_native_support()) {
+ snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+ }else if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
&& audio_extn_get_anc_enabled()) {
#ifdef RECORD_PLAY_CONCURRENCY
if (use_voip_out_devices) {
@@ -2766,6 +2897,7 @@
static void set_audiocal(void *platform, struct str_parms *parms, char *value, int len) {
struct platform_data *my_data = (struct platform_data *)platform;
+ struct stream_out out={0};
acdb_audio_cal_cfg_t cal={0};
uint8_t *dptr = NULL;
int32_t dlen;
@@ -2802,7 +2934,8 @@
if(audio_is_input_device(cal.dev_id)) {
cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
} else {
- cal.snd_dev_id = platform_get_output_snd_device(platform, cal.dev_id);
+ out.devices = cal.dev_id;
+ cal.snd_dev_id = platform_get_output_snd_device(platform, &out);
}
}
cal.acdb_dev_id = platform_get_snd_device_acdb_id(cal.snd_dev_id);
@@ -2895,6 +3028,7 @@
}
}
#endif
+ native_audio_set_params(platform, parms, value, sizeof(value));
ALOGV("%s: exit with code(%d)", __func__, ret);
return ret;
}
@@ -3001,6 +3135,7 @@
static void get_audiocal(void *platform, void *keys, void *pReply) {
struct platform_data *my_data = (struct platform_data *)platform;
+ struct stream_out out={0};
struct str_parms *query = (struct str_parms *)keys;
struct str_parms *reply=(struct str_parms *)pReply;
acdb_audio_cal_cfg_t cal={0};
@@ -3031,7 +3166,8 @@
if(cal.dev_id & AUDIO_DEVICE_BIT_IN) {
cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
} else if(cal.dev_id) {
- cal.snd_dev_id = platform_get_output_snd_device(platform, cal.dev_id);
+ out.devices = cal.dev_id;
+ cal.snd_dev_id = platform_get_output_snd_device(platform, &out);
}
cal.acdb_dev_id = platform_get_snd_device_acdb_id(cal.snd_dev_id);
if (cal.acdb_dev_id < 0) {
@@ -3127,6 +3263,7 @@
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VOLUME_BOOST, value);
}
+ native_audio_get_params(query, reply, value, sizeof(value));
/* Handle audio calibration keys */
kv_pairs = str_parms_to_str(reply);
@@ -3344,26 +3481,64 @@
return fragment_size;
}
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+ return OFFLOAD_USE_SMALL_BUFFER;
+}
+
+int platform_is_external_codec (char *snd_card_name)
+{
+
+ if (!strncmp(snd_card_name, "msm8952-tomtom-snd-card",
+ sizeof("msm8952-tomtom-snd-card")) ||
+ !strncmp(snd_card_name, "msm8976-tasha-snd-card",
+ sizeof("msm8976-tasha-snd-card")))
+ {
+ /* external codec, for rest/old of the external codecs
+ we dont support this funtionality(chaning AFE params)
+ at the monment
+ */
+ return 1;
+ }
+ else {
+ /* internal codec */
+ return 0;
+ }
+}
+
int platform_set_codec_backend_cfg(struct audio_device* adev,
+ snd_device_t snd_device,
unsigned int bit_width, unsigned int sample_rate)
{
- ALOGV("%s bit width: %d, sample rate: %d", __func__, bit_width, sample_rate);
-
int ret = 0;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+ ALOGV("%s bit width: %d, sample rate: %d\n", __func__, bit_width, sample_rate);
+
const char *snd_card_name = mixer_get_name(adev->mixer);
- if (bit_width != adev->cur_codec_backend_bit_width) {
- 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";
+ int is_external_codec = platform_is_external_codec(snd_card_name);
+
+
+
+ backend_idx = platform_get_backend_index(snd_device);
+ ALOGV("%s bit width: %d, sample rate: %d backend_idx - %d\n",
+ __func__, bit_width, sample_rate, backend_idx);
+
+ if (bit_width !=
+ my_data->current_backend_cfg[backend_idx].bit_width) {
+
struct mixer_ctl *ctl;
- ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!is_external_codec) {
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ "MI2S_RX Format");
+
+ } else {
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
+ }
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer command - %s",
- __func__, mixer_ctl_name);
+ __func__, my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
return -EINVAL;
}
@@ -3371,10 +3546,12 @@
mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ if (backend_idx != HEADPHONE_44_1_BACKEND)
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
- adev->cur_codec_backend_bit_width = bit_width;
- ALOGE("Backend bit width is set to %d ", bit_width);
+ my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
+ ALOGD("%s: %s mixer set to %d bit", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
}
/*
@@ -3386,9 +3563,12 @@
* Upper limit is inclusive in the sample rate range.
*/
// TODO: This has to be more dynamic based on policy file
- if (sample_rate != adev->cur_codec_backend_samplerate) {
+
+ if ((sample_rate != my_data->current_backend_cfg[(int)backend_idx].sample_rate) &&
+ (is_external_codec) ) {
+ /* sample rate update is needed only for external codecs which
+ support 24 bit playback*/
char *rate_str = NULL;
- const char * mixer_ctl_name = "SLIM_0_RX SampleRate";
struct mixer_ctl *ctl;
switch (sample_rate) {
@@ -3397,10 +3577,12 @@
case 16000:
case 22050:
case 32000:
- case 44100:
case 48000:
rate_str = "KHZ_48";
break;
+ case 44100:
+ rate_str = "KHZ_44P1";
+ break;
case 64000:
case 88200:
case 96000:
@@ -3415,31 +3597,46 @@
break;
}
- ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_backend_cfg[backend_idx].samplerate_mixer_ctl);
if(!ctl) {
ALOGE("%s: Could not get ctl for mixer command - %s",
- __func__, mixer_ctl_name);
+ __func__, my_data->current_backend_cfg[backend_idx].samplerate_mixer_ctl);
return -EINVAL;
}
- ALOGV("Set sample rate as rate_str = %s", rate_str);
+ ALOGD("%s: %s set to %s", __func__,
+ my_data->current_backend_cfg[backend_idx].samplerate_mixer_ctl, rate_str);
mixer_ctl_set_enum_by_string(ctl, rate_str);
- adev->cur_codec_backend_samplerate = sample_rate;
+ my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
}
return ret;
}
bool platform_check_codec_backend_cfg(struct audio_device* adev,
- struct audio_usecase* usecase __unused,
+ struct audio_usecase* usecase,
+ snd_device_t snd_device,
unsigned int* new_bit_width,
unsigned int* new_sample_rate)
{
bool backend_change = false;
struct listnode *node;
struct stream_out *out = NULL;
- unsigned int bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
- unsigned int sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ unsigned int bit_width;
+ unsigned int sample_rate;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+
+ backend_idx = platform_get_backend_index(snd_device);
+
+ bit_width = *new_bit_width;
+ sample_rate = *new_sample_rate;
+
+ ALOGI("%s Codec selected backend: %d current bit width: %d and sample rate: %d",
+ __func__, backend_idx, bit_width, sample_rate);
// For voice calls use default configuration
// force routing is not required here, caller will do it anyway
@@ -3460,36 +3657,50 @@
list_for_each(node, &adev->usecase_list) {
struct audio_usecase *curr_usecase;
curr_usecase = node_to_item(node, struct audio_usecase, list);
- if (curr_usecase->type == PCM_PLAYBACK) {
+ if (curr_usecase->type == PCM_PLAYBACK &&
+ usecase != curr_usecase) {
struct stream_out *out =
(struct stream_out*) curr_usecase->stream.out;
- if (out != NULL ) {
- ALOGV("Offload playback running bw %d sr %d",
- out->bit_width, out->sample_rate);
+ usecase_backend_idx = platform_get_backend_index(curr_usecase->out_snd_device);
+
+ if (out != NULL &&
+ usecase_backend_idx == backend_idx) {
+ ALOGV("%s: usecase Offload playback running bw %d sr %d device %s be_idx %d",
+ __func__, out->bit_width, out->sample_rate,
+ platform_get_snd_device_name(curr_usecase->out_snd_device), usecase_backend_idx);
if (bit_width < out->bit_width)
bit_width = out->bit_width;
if (sample_rate < out->sample_rate)
sample_rate = out->sample_rate;
+ if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
}
}
}
+ if (backend_idx != HEADPHONE_44_1_BACKEND) {
+ // 16 bit playbacks are allowed through 16 bit/48 khz backend only for
+ // all non-native streams
+ if (16 == bit_width) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s: resetting sample_rate back to default, "
+ "backend_idx: %d", __func__, backend_idx);
+ }
- // 16 bit playback on speakers is allowed through 48 khz backend only
- if (16 == bit_width) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ // 24 bit playback on speakers is allowed through 48 khz backend only
+ // bit width re-configured based on platform info
+ if ((24 == bit_width) &&
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
+ bit_width = (uint32_t)platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ }
}
- // 24 bit playback on speakers is allowed through 48 khz backend only
- // bit width re-configured based on platform info
- if ((24 == bit_width) &&
- (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
- bit_width = (uint32_t)platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- }
+ ALOGI("%s Codec selected backend: %d updated bit width: %d and sample rate: %d",
+ __func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
- if ((bit_width != adev->cur_codec_backend_bit_width) ||
- (sample_rate != adev->cur_codec_backend_samplerate)) {
+ if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
*new_bit_width = bit_width;
*new_sample_rate = sample_rate;
backend_change = true;
@@ -3501,21 +3712,26 @@
}
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
- struct audio_usecase *usecase,
- snd_device_t snd_device __unused)
+ struct audio_usecase *usecase, snd_device_t snd_device)
{
- ALOGV("platform_check_and_set_codec_backend_cfg usecase = %d",usecase->id );
+ unsigned int new_bit_width;
+ unsigned int new_sample_rate;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
- unsigned int new_bit_width, old_bit_width;
- unsigned int new_sample_rate, old_sample_rate;
+ ALOGV("%s: usecase = %d", __func__, usecase->id );
- new_bit_width = old_bit_width = adev->cur_codec_backend_bit_width;
- new_sample_rate = old_sample_rate = adev->cur_codec_backend_samplerate;
+ backend_idx = platform_get_backend_index(snd_device);
- ALOGW("Codec backend bitwidth %d, samplerate %d", old_bit_width, old_sample_rate);
- if (platform_check_codec_backend_cfg(adev, usecase,
+ new_bit_width = usecase->stream.out->bit_width;
+ new_sample_rate = usecase->stream.out->sample_rate;
+
+ ALOGI("%s: Usecase bitwidth %d, samplerate %d, backend_idx %d",
+ __func__, new_bit_width, new_sample_rate, backend_idx);
+ if (platform_check_codec_backend_cfg(adev, usecase, snd_device,
&new_bit_width, &new_sample_rate)) {
- platform_set_codec_backend_cfg(adev, new_bit_width, new_sample_rate);
+ platform_set_codec_backend_cfg(adev, snd_device,
+ new_bit_width, new_sample_rate);
return true;
}
@@ -4182,3 +4398,9 @@
done:
return ret;
}
+
+bool platform_send_gain_dep_cal(void *platform __unused,
+ int level __unused)
+{
+ return 0;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 3ce4c57..c63bfe9 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -37,7 +37,11 @@
/*
* Below are the devices for which is back end is same, SLIMBUS_0_RX.
* All these devices are handled by the internal HW codec. We can
- * enable any one of these devices at any time
+ * enable any one of these devices at any time. An exception here is
+ * 44.1k headphone which uses different backend. This is filtered
+ * as different hal internal device in the code but remains same
+ * as standard android device AUDIO_DEVICE_OUT_WIRED_HEADPHONE
+ * for other layers.
*/
#define AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND \
(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
@@ -163,13 +167,16 @@
};
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
-#define OUTPUT_SAMPLING_RATE_44100 44100
-
+#define OUTPUT_SAMPLING_RATE_44100 44100
+#define MAX_PORT 6
+#define ALL_CODEC_BACKEND_PORT 0
+#define HEADPHONE_44_1_BACKEND_PORT 5
enum {
DEFAULT_CODEC_BACKEND,
HEADPHONE_44_1_BACKEND,
MAX_CODEC_BACKENDS
};
+#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO "audio.nat.codec.enabled"
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION_MS 20
@@ -189,8 +196,8 @@
* We should take care of returning proper size when AudioFlinger queries for
* the buffer size of an input/output stream
*/
-#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
-#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 5
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 5fab099..861790e 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1098,6 +1098,11 @@
return 0;
}
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+ return false;
+}
+
int platform_get_edid_info(void *platform __unused)
{
return -ENOSYS;
@@ -1166,3 +1171,9 @@
{
return -ENOSYS;
}
+
+bool platform_send_gain_dep_cal(void *platform __unused,
+ int level __unused)
+{
+ return 0;
+}
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index c96d11e..7849644 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -134,6 +134,10 @@
SND_DEVICE_IN_HANDSET_MIC_EXTERNAL
};
+static const snd_device_t tasha_DB_variant_devices[] = {
+ SND_DEVICE_OUT_SPEAKER
+};
+
static const snd_device_t taiko_apq8084_sbc_variant_devices[] = {
SND_DEVICE_IN_HANDSET_MIC,
SND_DEVICE_IN_SPEAKER_MIC,
@@ -278,6 +282,12 @@
hw_info->snd_devices = (snd_device_t *)tomtom_liquid_variant_devices;
hw_info->num_snd_devices = ARRAY_SIZE(tomtom_liquid_variant_devices);
strlcpy(hw_info->dev_extn, "-liquid", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8996-tasha-db-snd-card")) {
+ strlcpy(hw_info->type, " dragon-board", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
+ hw_info->snd_devices = (snd_device_t *)tasha_DB_variant_devices;
+ hw_info->num_snd_devices = ARRAY_SIZE(tasha_DB_variant_devices);
+ strlcpy(hw_info->dev_extn, "-db", sizeof(hw_info->dev_extn));
} else {
ALOGW("%s: Not a 8996 device", __func__);
}
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index d9921d0..875cc2c 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -54,6 +54,7 @@
#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/system/etc/audio_platform_info_i2s.xml"
+#include <linux/msm_audio.h>
#define LIB_ACDB_LOADER "libacdbloader.so"
#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
@@ -129,7 +130,6 @@
char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
[WCD9XXX_ANC_CAL] = "anc_cal",
[WCD9XXX_MBHC_CAL] = "mbhc_cal",
- [WCD9XXX_MAD_CAL] = "mad_cal",
};
enum {
@@ -156,6 +156,12 @@
uint32_t param_id;
} acdb_audio_cal_cfg_t;
+enum {
+ CAL_MODE_SEND = 0x1,
+ CAL_MODE_PERSIST = 0x2,
+ CAL_MODE_RTAC = 0x4
+};
+
/* Audio calibration related functions */
typedef void (*acdb_deallocate_t)();
typedef int (*acdb_init_t)(const char *, char *, int);
@@ -181,6 +187,7 @@
bool ui_na_prop_enabled;
} native_audio_prop;
static native_audio_prop na_props = {0, 0};
+typedef int (*acdb_send_gain_dep_cal_t)(int, int, int, int, int);
struct platform_data {
struct audio_device *adev;
@@ -213,6 +220,7 @@
acdb_send_common_top_t acdb_send_common_top;
void *hw_info;
+ acdb_send_gain_dep_cal_t acdb_send_gain_dep_cal;
struct csd_data *csd;
void *edid_info;
bool edid_valid;
@@ -708,6 +716,55 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+bool platform_send_gain_dep_cal(void *platform, int level) {
+ bool ret_val = false;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct audio_device *adev = my_data->adev;
+ int acdb_dev_id, app_type;
+ int acdb_dev_type = MSM_SNDDEV_CAP_RX;
+ int mode = CAL_MODE_RTAC;
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ if (my_data->acdb_send_gain_dep_cal == NULL) {
+ ALOGE("%s: dlsym error for acdb_send_gain_dep_cal", __func__);
+ return ret_val;
+ }
+
+ if (!voice_is_in_call(adev)) {
+ ALOGV("%s: Not Voice call usecase, apply new cal for level %d",
+ __func__, level);
+
+ // find the current active sound device
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+
+ if (usecase != NULL &&
+ usecase->type == PCM_PLAYBACK &&
+ (usecase->stream.out->devices == AUDIO_DEVICE_OUT_SPEAKER ||
+ usecase->stream.out->devices == AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
+
+ ALOGV("%s: out device is %d", __func__, usecase->out_snd_device);
+ app_type = usecase->stream.out->app_type_cfg.app_type;
+ acdb_dev_id = acdb_device_table[usecase->out_snd_device];
+ if (!my_data->acdb_send_gain_dep_cal(acdb_dev_id, app_type,
+ acdb_dev_type, mode, level)) {
+ // set ret_val true if at least one calibration is set successfully
+ ret_val = true;
+ } else {
+ ALOGE("%s: my_data->acdb_send_gain_dep_cal failed ", __func__);
+ }
+ } else {
+ ALOGW("%s: Usecase list is empty", __func__);
+ }
+ }
+ } else {
+ ALOGW("%s: Voice call in progress .. ignore setting new cal",
+ __func__);
+ }
+ return ret_val;
+}
+
void platform_set_echo_reference(void *platform, bool enable)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -991,8 +1048,6 @@
struct wcdcal_ioctl_buffer codec_buffer;
struct param_data calib;
- if (!strcmp(cal_name_info[type], "mad_cal"))
- calib.acdb_id = SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID;
calib.get_size = 1;
ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
&calib);
@@ -1289,6 +1344,12 @@
ALOGE("%s: Could not find the symbol acdb_get_default_app_type from %s",
__func__, LIB_ACDB_LOADER);
+ my_data->acdb_send_gain_dep_cal = (acdb_send_gain_dep_cal_t)dlsym(my_data->acdb_handle,
+ "acdb_loader_send_gain_dep_cal");
+ if (!my_data->acdb_send_gain_dep_cal)
+ ALOGV("%s: Could not find the symbol acdb_loader_send_gain_dep_cal from %s",
+ __func__, LIB_ACDB_LOADER);
+
my_data->acdb_send_common_top = (acdb_send_common_top_t)dlsym(
my_data->acdb_handle,
"acdb_loader_send_common_custom_topology");
@@ -3249,6 +3310,11 @@
return fragment_size;
}
+bool platform_use_small_buffer(audio_offload_info_t* info)
+{
+ return OFFLOAD_USE_SMALL_BUFFER;
+}
+
int platform_set_codec_backend_cfg(struct audio_device* adev,
snd_device_t snd_device,
unsigned int bit_width, unsigned int sample_rate)
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 4b90f8b..2247933 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -185,8 +185,8 @@
* We should take care of returning proper size when AudioFlinger queries for
* the buffer size of an input/output stream
*/
-#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
-#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 5
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
@@ -227,6 +227,8 @@
#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
#ifdef PLATFORM_APQ8084
#define SPKR_PROT_CALIB_TX_PCM_DEVICE 35
+#elif PLATFORM_MSM8996
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 42
#else
#define SPKR_PROT_CALIB_TX_PCM_DEVICE 25
#endif
diff --git a/hal/platform_api.h b/hal/platform_api.h
index d9fe0c0..921ff0b 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -30,6 +30,7 @@
int platform_get_snd_device_name_extn(void *platform, snd_device_t snd_device,
char *device_name);
void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
+bool platform_send_gain_dep_cal(void *platform, int level);
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
int platform_get_snd_device_index(char *snd_device_index_name);
int platform_set_fluence_type(void *platform, char *value);
@@ -94,6 +95,7 @@
struct audio_offload_info_t;
uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
+bool platform_use_small_buffer(audio_offload_info_t* info);
uint32_t platform_get_compress_passthrough_buffer_size(audio_offload_info_t* info);
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
diff --git a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
index 6af9269..4cfee1b 100644
--- a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
+++ b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
@@ -4208,6 +4208,16 @@
buffer->nOffset = 0;
buffer->nTimeStamp = nTimestamp;
frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
+
+ if (errno == ENETRESET)
+ {
+ ALOGE("In SSR, return error to close the session");
+ m_cb.EventHandler(&m_cmp,
+ m_app_data,
+ OMX_EventError,
+ OMX_ErrorHardware,
+ 0, NULL );
+ }
return OMX_ErrorNone;
}
}
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 71a210f..192320a 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -14,8 +14,12 @@
virtualizer.c \
reverb.c \
effect_api.c \
- effect_util.c \
- hw_accelerator.c
+ effect_util.c
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS)),true)
+ LOCAL_CFLAGS += -DHW_ACCELERATED_EFFECTS
+ LOCAL_SRC_FILES += hw_accelerator.c
+endif
LOCAL_CFLAGS+= -O2 -fvisibility=hidden
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index e38a41c..6f1bdfa 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -38,6 +38,7 @@
#define LOG_TAG "offload_effect_bundle"
//#define LOG_NDEBUG 0
+#include <stdlib.h>
#include <cutils/list.h>
#include <cutils/log.h>
#include <system/thread_defs.h>
@@ -69,7 +70,9 @@
&ins_env_reverb_descriptor,
&aux_preset_reverb_descriptor,
&ins_preset_reverb_descriptor,
+#ifdef HW_ACCELERATED_EFFECTS
&hw_accelerator_descriptor,
+#endif
NULL,
};
@@ -302,9 +305,6 @@
goto exit;
}
- if (out_ctxt->mixer)
- mixer_close(out_ctxt->mixer);
-
list_for_each(fx_node, &out_ctxt->effects_list) {
effect_context_t *fx_ctxt = node_to_item(fx_node,
effect_context_t,
@@ -313,6 +313,9 @@
fx_ctxt->ops.stop(fx_ctxt, out_ctxt);
}
+ if (out_ctxt->mixer)
+ mixer_close(out_ctxt->mixer);
+
list_remove(&out_ctxt->outputs_list_node);
#ifdef DTS_EAGLE
@@ -604,6 +607,7 @@
reverb_preset_init(reverb_ctxt);
}
reverb_ctxt->ctl = NULL;
+#ifdef HW_ACCELERATED_EFFECTS
} else if (memcmp(uuid, &hw_accelerator_descriptor.uuid,
sizeof(effect_uuid_t)) == 0) {
hw_accelerator_context_t *hw_acc_ctxt = (hw_accelerator_context_t *)
@@ -625,6 +629,7 @@
context->ops.process = hw_accelerator_process;
context->desc = &hw_accelerator_descriptor;
+#endif
} else {
return -EINVAL;
}
@@ -911,6 +916,7 @@
} break;
+#ifdef HW_ACCELERATED_EFFECTS
case EFFECT_CMD_HW_ACC: {
ALOGV("EFFECT_CMD_HW_ACC cmdSize %d pCmdData %p, *replySize %d, pReplyData %p",
cmdSize, pCmdData, *replySize, pReplyData);
@@ -925,6 +931,7 @@
context->hw_acc_enabled = (value > 0) ? true : false;
break;
}
+#endif
default:
if (cmdCode >= EFFECT_CMD_FIRST_PROPRIETARY && context->ops.command)
status = context->ops.command(context, cmdCode, cmdSize,
diff --git a/post_proc/effect_api.c b/post_proc/effect_api.c
index e15db17..16c370b 100644
--- a/post_proc/effect_api.c
+++ b/post_proc/effect_api.c
@@ -55,6 +55,7 @@
#endif
#include <stdbool.h>
+#include <errno.h>
#include <cutils/log.h>
#include <tinyalsa/asoundlib.h>
#include <sound/audio_effects.h>
diff --git a/post_proc/effect_util.c b/post_proc/effect_util.c
index 8f7a604..52f623d 100644
--- a/post_proc/effect_util.c
+++ b/post_proc/effect_util.c
@@ -16,6 +16,7 @@
#include <utils/Log.h>
#include <stdlib.h>
+#include <string.h>
#include "effect_util.h"
#ifdef LOG_TAG
diff --git a/voice_processing/voice_processing.c b/voice_processing/voice_processing.c
index 1d18a3d..72e4d84 100644
--- a/voice_processing/voice_processing.c
+++ b/voice_processing/voice_processing.c
@@ -16,6 +16,7 @@
#define LOG_TAG "voice_processing"
/*#define LOG_NDEBUG 0*/
+#include <stdlib.h>
#include <dlfcn.h>
#include <cutils/log.h>
#include <cutils/list.h>