Merge commit 'eb9127a942289de0cf9587d7456cc6c9a44eb411' into remote_branch
Change-Id: Ic630ae55085197a1bc7b79c726f7a0e39b9927b4
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 523a875..8e5d64e 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -428,7 +428,6 @@
audio_format_t format,
uint32_t sample_rate,
uint32_t bit_width,
- audio_channel_mask_t channel_mask,
struct stream_app_type_cfg *app_type_cfg);
int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase);
void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
diff --git a/hal/audio_extn/dts_eagle.c b/hal/audio_extn/dts_eagle.c
index 52d7abb..7608a61 100644
--- a/hal/audio_extn/dts_eagle.c
+++ b/hal/audio_extn/dts_eagle.c
@@ -39,7 +39,6 @@
#define STATE_NOTIFY_FILE "/data/misc/dts/stream"
#define FADE_NOTIFY_FILE "/data/misc/dts/fade"
#define DTS_EAGLE_KEY "DTS_EAGLE"
-#define DEVICE_NODE "/dev/snd/hwC0D3"
#define MAX_LENGTH_OF_INTEGER_IN_STRING 13
#define PARAM_GET_MAX_SIZE 512
@@ -117,7 +116,7 @@
}
if (!sent) {
- int fd = open(DEVICE_NODE, O_RDWR);
+ int fd = open(GENERIC_AUDIO_DEVICE_NODE, O_RDWR);
if (get) {
ALOGD("DTS_EAGLE_HAL (%s): no stream opened, attempting to retrieve directly from cache", __func__);
@@ -137,7 +136,7 @@
}
close(fd);
} else {
- ALOGE("DTS_EAGLE_HAL (%s): couldn't open device %s\n", __func__, DEVICE_NODE);
+ ALOGE("DTS_EAGLE_HAL (%s): couldn't open device %s\n", __func__, GENERIC_AUDIO_DEVICE_NODE);
ret = -EINVAL;
}
}
diff --git a/hal/audio_extn/fm.c b/hal/audio_extn/fm.c
index ed3776c..efc2307 100644
--- a/hal/audio_extn/fm.c
+++ b/hal/audio_extn/fm.c
@@ -243,7 +243,20 @@
if (ret >= 0) {
val = atoi(value);
if(val > 0)
- select_devices(adev, USECASE_AUDIO_PLAYBACK_FM);
+ /*
+ * Only when wsa does present and is in analog mode,
+ * fm will stop/start here.
+ * To-do: when the kernel codec type query change
+ * is ready, enum of wsa mode should be checked here.
+ * Currently, platform_get_wsa_mode will directly return
+ * 1 when wsa is in analog mode.
+ */
+ if (platform_get_wsa_mode(adev->platform) == 1) {
+ fm_stop(adev);
+ fm_start(adev);
+ } else {
+ select_devices(adev, USECASE_AUDIO_PLAYBACK_FM);
+ }
}
}
if (fmmod.restart_fm && (fmmod.scard_state == SND_CARD_STATE_ONLINE)) {
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 15f2a3a..e5201cc 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -779,7 +779,6 @@
__func__, NUM_ATTEMPTS);
t0_spk_1 = SAFE_SPKR_TEMP_Q6;
}
- fclose(fp);
}
if (spk_2_tzn > 0) {
snprintf(wsa_path, MAX_PATH, TZ_WSA, spk_2_tzn);
@@ -810,7 +809,6 @@
__func__, NUM_ATTEMPTS);
t0_spk_2 = SAFE_SPKR_TEMP_Q6;
}
- fclose(fp);
}
}
pthread_mutex_unlock(&adev->lock);
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index dbd54f7..75bb336 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -436,6 +436,7 @@
* than all sample rates in list for the input bit width.
*/
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+
list_for_each(node_i, &so_info->sample_rate_list) {
ss_info = node_to_item(node_i, struct stream_sample_rate, list);
if ((sample_rate <= ss_info->sample_rate) &&
@@ -443,7 +444,7 @@
app_type_cfg->app_type = so_info->app_type_cfg.app_type;
app_type_cfg->sample_rate = sample_rate;
app_type_cfg->bit_width = so_info->app_type_cfg.bit_width;
- ALOGV("%s Assuming default sample rate. app_type_cfg->app_type %d, app_type_cfg->sample_rate %d, app_type_cfg->bit_width %d",
+ ALOGV("%s Assuming sample rate. app_type_cfg->app_type %d, app_type_cfg->sample_rate %d, app_type_cfg->bit_width %d",
__func__, app_type_cfg->app_type, app_type_cfg->sample_rate, app_type_cfg->bit_width);
return true;
}
@@ -458,14 +459,12 @@
audio_format_t format,
uint32_t sample_rate,
uint32_t bit_width,
- audio_channel_mask_t channel_mask,
struct stream_app_type_cfg *app_type_cfg)
{
struct listnode *node_i, *node_j, *node_k;
struct streams_output_cfg *so_info;
struct stream_format *sf_info;
struct stream_sample_rate *ss_info;
- char value[PROPERTY_VALUE_MAX] = {0};
if ((24 == bit_width) &&
(devices & AUDIO_DEVICE_OUT_SPEAKER)) {
@@ -476,16 +475,6 @@
ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
}
- property_get("audio.playback.mch.downsample",value,"");
- if (!strncmp("true", value, sizeof("true"))) {
- if ((popcount(channel_mask) > 2) &&
- (sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
- !(flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s: MCH session defaulting sample rate to %d",
- __func__, sample_rate);
- }
- }
ALOGV("%s: flags: %x, format: %x sample_rate %d",
__func__, flags, format, sample_rate);
list_for_each(node_i, streams_output_cfg_list) {
@@ -527,7 +516,6 @@
struct mixer_ctl *ctl;
int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
- char value[PROPERTY_VALUE_MAX] = {0};
ALOGV("%s", __func__);
@@ -573,18 +561,14 @@
if ((24 == usecase->stream.out->bit_width) &&
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ } else if ((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
+ usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||
+ (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
+ sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
} else {
sample_rate = out->app_type_cfg.sample_rate;
}
- property_get("audio.playback.mch.downsample",value,"");
- if (!strncmp("true", value, sizeof("true"))) {
- if ((popcount(out->channel_mask) > 2) &&
- (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
- !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- }
-
app_type_cfg[len++] = out->app_type_cfg.app_type;
app_type_cfg[len++] = acdb_dev_id;
if (((out->format == AUDIO_FORMAT_E_AC3) ||
@@ -593,6 +577,7 @@
app_type_cfg[len++] = sample_rate * 4;
else
app_type_cfg[len++] = sample_rate;
+
mixer_ctl_set_array(ctl, app_type_cfg, len);
ALOGI("%s app_type %d, acdb_dev_id %d, sample_rate %d",
__func__, out->app_type_cfg.app_type, acdb_dev_id, sample_rate);
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 7c3848b..126f3b0 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -582,6 +582,8 @@
struct audio_usecase *usecase;
bool switch_device[AUDIO_USECASE_MAX];
int i, num_uc_to_switch = 0;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
/*
* This function is to make sure that all the usecases that are active on
@@ -598,23 +600,34 @@
* If there is a backend configuration change for the device when a
* new stream starts, then ADM needs to be closed and re-opened with the new
* configuraion. This call check if we need to re-route all the streams
- * associated with the backend. Touch tone + 24 bit playback.
+ * associated with the backend. Touch tone + 24 bit + native playback.
*/
- bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info);
-
+ bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info,
+ snd_device);
+ backend_idx = platform_get_backend_index(snd_device);
/* Disable all the usecases on the shared backend other than the
- specified usecase */
+ * specified usecase.
+ */
for (i = 0; i < AUDIO_USECASE_MAX; i++)
switch_device[i] = false;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
+
+ if (usecase == uc_info)
+ continue;
+ usecase_backend_idx = platform_get_backend_index(usecase->out_snd_device);
+ 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));
+
if (usecase->type != PCM_CAPTURE &&
- usecase != uc_info &&
(usecase->out_snd_device != snd_device || force_routing) &&
- usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
- ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
- __func__, use_case_table[usecase->id],
+ usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&
+ usecase_backend_idx == backend_idx) {
+ ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", __func__,
+ use_case_table[usecase->id],
platform_get_snd_device_name(usecase->out_snd_device));
disable_audio_route(adev, usecase);
switch_device[usecase->id] = true;
@@ -814,7 +827,7 @@
(usecase->type == VOIP_CALL) ||
(usecase->type == PCM_HFP_CALL)) {
out_snd_device = platform_get_output_snd_device(adev->platform,
- usecase->stream.out->devices);
+ usecase->stream.out);
in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
usecase->devices = usecase->stream.out->devices;
} else {
@@ -854,7 +867,7 @@
in_snd_device = SND_DEVICE_NONE;
if (out_snd_device == SND_DEVICE_NONE) {
out_snd_device = platform_get_output_snd_device(adev->platform,
- usecase->stream.out->devices);
+ usecase->stream.out);
if (usecase->stream.out == adev->primary_output &&
adev->active_input &&
adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
@@ -899,6 +912,9 @@
(usecase->in_snd_device != SND_DEVICE_NONE) &&
(usecase->out_snd_device != SND_DEVICE_NONE)) {
status = platform_switch_voice_call_device_pre(adev->platform);
+ /* Disable sidetone only if voice call already exists */
+ if (voice_is_call_state_active(adev))
+ voice_set_sidetone(adev, usecase->out_snd_device, false);
}
/* Disable current sound devices */
@@ -941,6 +957,9 @@
out_snd_device,
in_snd_device);
enable_audio_route_for_voice_usecases(adev, usecase);
+ /* Enable sidetone only if voice call already exists */
+ if (voice_is_call_state_active(adev))
+ voice_set_sidetone(adev, out_snd_device, true);
}
usecase->in_snd_device = in_snd_device;
@@ -954,7 +973,6 @@
usecase->stream.out->format,
usecase->stream.out->sample_rate,
usecase->stream.out->bit_width,
- usecase->stream.out->channel_mask,
&usecase->stream.out->app_type_cfg);
ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
}
@@ -2979,8 +2997,7 @@
audio_extn_utils_update_stream_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
devices, flags, format, out->sample_rate,
- out->bit_width, out->channel_mask,
- &out->app_type_cfg);
+ out->bit_width, &out->app_type_cfg);
if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
(flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
/* Ensure the default output is not selected twice */
@@ -3135,6 +3152,12 @@
} else if (strstr(snd_card_status, "ONLINE")) {
ALOGD("Received sound card ONLINE status");
set_snd_card_state(adev,SND_CARD_STATE_ONLINE);
+ if (!platform_is_acdb_initialized(adev->platform)) {
+ ret = platform_acdb_init(adev->platform);
+ if(ret)
+ ALOGE("acdb initialization is failed");
+
+ }
}
}
@@ -3640,8 +3663,6 @@
adev->bluetooth_nrec = true;
adev->acdb_settings = TTY_MODE_OFF;
/* adev->cur_hdmi_channels = 0; by calloc() */
- adev->cur_codec_backend_samplerate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- adev->cur_codec_backend_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
voice_init(adev);
list_init(&adev->usecase_list);
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index 1a882db..fb36a95 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -245,6 +245,12 @@
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, "msm8976-tasha-skun-snd-card")) {
+ strlcpy(hw_info->type, "", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8976", 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));
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 765aea1..dfdcf7d 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -58,6 +58,7 @@
#define MIXER_XML_PATH_WCD9306 "/system/etc/mixer_paths_wcd9306.xml"
#define MIXER_XML_PATH_WCD9330 "/system/etc/mixer_paths_wcd9330.xml"
#define MIXER_XML_PATH_WCD9335 "/system/etc/mixer_paths_wcd9335.xml"
+#define MIXER_XML_PATH_SKUN "/system/etc/mixer_paths_qrd_skun.xml"
#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
#define PLATFORM_INFO_XML_PATH_EXTCODEC "/system/etc/audio_platform_info_extcodec.xml"
@@ -70,17 +71,14 @@
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
/* Used in calculating fragment size for pcm offload */
-#define PCM_OFFLOAD_BUFFER_DURATION_FOR_AV 1000 /* 1 sec */
-#define PCM_OFFLOAD_BUFFER_DURATION_FOR_AV_STREAMING 80 /* 80 millisecs */
-#define PCM_OFFLOAD_BUFFER_DURATION_FOR_SMALL_BUFFERS 20 /* 20 millisecs */
-#define PCM_OFFLOAD_BUFFER_DURATION_MAX 1200 /* 1200 millisecs */
+#define PCM_OFFLOAD_BUFFER_DURATION 40 /* 40 millisecs */
/* MAX PCM fragment size cannot be increased further due
* to flinger's cblk size of 1mb,and it has to be a multiple of
* 24 - lcm of channels supported by DSP
*/
#define MAX_PCM_OFFLOAD_FRAGMENT_SIZE (240 * 1024)
-#define MIN_PCM_OFFLOAD_FRAGMENT_SIZE (4 * 1024)
+#define MIN_PCM_OFFLOAD_FRAGMENT_SIZE 512
/*
* Offload buffer size for compress passthrough
@@ -192,6 +190,20 @@
typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
+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;
@@ -208,6 +220,7 @@
bool hd_voice;
bool ec_ref_enabled;
bool is_wsa_speaker;
+ bool is_acdb_initialized;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -226,6 +239,7 @@
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;
@@ -311,6 +325,7 @@
[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",
@@ -415,6 +430,7 @@
[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,
@@ -521,6 +537,7 @@
{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)},
@@ -728,7 +745,9 @@
!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")))
+ sizeof("msm8976-tasha-snd-card")) ||
+ !strncmp(snd_card_name, "msm8976-tasha-skun-snd-card",
+ sizeof("msm8976-tasha-skun-snd-card")))
{
ALOGI("%s: snd_card_name: %s",__func__,snd_card_name);
is_external_codec = true;
@@ -832,6 +851,14 @@
msm_be_id_array_len =
sizeof(msm_device_to_be_id_external_codec) / sizeof(msm_device_to_be_id_external_codec[0]);
+ } else if (!strncmp(snd_card_name, "msm8976-tasha-skun-snd-card",
+ sizeof("msm8976-tasha-skun-snd-card"))) {
+ strlcpy(mixer_xml_path, MIXER_XML_PATH_SKUN,
+ sizeof(MIXER_XML_PATH_SKUN));
+ 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, "msm8909-skua-snd-card",
sizeof("msm8909-skua-snd-card"))) {
strlcpy(mixer_xml_path, MIXER_XML_PATH_SKUA,
@@ -1075,7 +1102,7 @@
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.
@@ -1097,6 +1124,7 @@
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)
@@ -1219,6 +1247,37 @@
ALOGE("%s: Could not send anc cal", __FUNCTION__);
}
+int platform_acdb_init(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ char *cvd_version = NULL;
+ int key = 0;
+ const char *snd_card_name;
+ int result;
+ char value[PROPERTY_VALUE_MAX];
+ cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
+ if (!cvd_version)
+ ALOGE("Failed to allocate cvd version");
+ else
+ get_cvd_version(cvd_version, my_data->adev);
+
+ property_get("audio.ds1.metainfo.key",value,"0");
+ key = atoi(value);
+ snd_card_name = mixer_get_name(my_data->adev->mixer);
+ result = my_data->acdb_init(snd_card_name, cvd_version, key);
+ if (cvd_version)
+ free(cvd_version);
+ if (!result) {
+ my_data->is_acdb_initialized = true;
+ ALOGD("ACDB initialized");
+ audio_hwdep_send_cal(my_data);
+ } else {
+ my_data->is_acdb_initialized = false;
+ ALOGD("ACDB initialization failed");
+ }
+ return result;
+}
+
void *platform_init(struct audio_device *adev)
{
char platform[PROPERTY_VALUE_MAX];
@@ -1231,6 +1290,7 @@
char *cvd_version = NULL;
const char *mixer_ctl_name = "Set HPX ActiveBe";
struct mixer_ctl *ctl = NULL;
+ int idx;
my_data = calloc(1, sizeof(struct platform_data));
if (!my_data) {
@@ -1349,8 +1409,6 @@
acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 131;
acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131;
}
- property_get("audio.ds1.metainfo.key",value,"0");
- key = atoi(value);
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
@@ -1407,15 +1465,7 @@
ALOGE("%s: dlsym error %s for acdb_loader_init_v2", __func__, dlerror());
goto acdb_init_fail;
}
-
- cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
- if (!cvd_version)
- ALOGE("Failed to allocate cvd version");
- else
- get_cvd_version(cvd_version, adev);
- my_data->acdb_init(snd_card_name, cvd_version, key);
- if (cvd_version)
- free(cvd_version);
+ platform_acdb_init(my_data);
}
audio_extn_pm_vote();
@@ -1474,6 +1524,26 @@
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;
}
@@ -1511,6 +1581,13 @@
audio_extn_dap_hal_deinit();
}
+int platform_is_acdb_initialized(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ ALOGD("%s: acdb initialized %d\n", __func__, my_data->is_acdb_initialized);
+ return my_data->is_acdb_initialized;
+}
+
const char *platform_get_snd_device_name(snd_device_t snd_device)
{
if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX)
@@ -1730,6 +1807,109 @@
return backend_bit_width_table[snd_device];
}
+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_native_support()
+{
+ 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,
int app_type, int sample_rate)
{
@@ -2016,12 +2196,14 @@
return ret;
}
-snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
+snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)
{
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
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;
@@ -2046,7 +2228,7 @@
goto exit;
}
- if (popcount(devices) == 2) {
+ if (popcount(devices) == 2 && !voice_is_in_call(adev)) {
if (devices == (AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
AUDIO_DEVICE_OUT_SPEAKER)) {
if (my_data->external_spk_1)
@@ -2085,7 +2267,8 @@
goto exit;
}
- if (voice_is_in_call(adev) || voice_extn_compress_voip_is_active(adev)) {
+ if ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_extn_compress_voip_is_active(adev)) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
@@ -2142,7 +2325,10 @@
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) {
@@ -2234,8 +2420,8 @@
ALOGV("%s: enter: out_device(%#x) in_device(%#x)",
__func__, out_device, in_device);
if (my_data->external_mic) {
- if ((out_device != AUDIO_DEVICE_NONE && voice_is_in_call(adev)) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev)) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_EARPIECE ||
out_device & AUDIO_DEVICE_OUT_SPEAKER )
@@ -2249,7 +2435,7 @@
if (snd_device != AUDIO_DEVICE_NONE)
goto exit;
- if ((out_device != AUDIO_DEVICE_NONE) && ((voice_is_in_call(adev)) ||
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
!voice_extn_compress_voip_is_active(adev)) {
@@ -2756,6 +2942,7 @@
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;
@@ -2792,7 +2979,8 @@
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);
@@ -2885,6 +3073,7 @@
}
}
#endif
+ native_audio_set_params(platform, parms, value, sizeof(value));
ALOGV("%s: exit with code(%d)", __func__, ret);
return ret;
}
@@ -2991,6 +3180,7 @@
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};
@@ -3021,7 +3211,8 @@
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) {
@@ -3117,6 +3308,7 @@
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VOLUME_BOOST, value);
}
+ native_audio_get_params(query, reply, value, sizeof(value));
ret = str_parms_get_str(query, AUDIO_PARAMETER_IS_HW_DECODER_SESSION_ALLOWED,
value, sizeof(value));
@@ -3339,25 +3531,13 @@
{
uint32_t fragment_size = 0;
uint32_t bits_per_sample = 16;
- uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_SMALL_BUFFERS;
+ uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
bits_per_sample = 32;
}
- if (platform_use_small_buffer(info)) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_SMALL_BUFFERS;
- } else {
- if (!info->has_video) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_MAX;
- } else if (info->has_video && info->is_streaming) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_AV_STREAMING;
- } else if (info->has_video) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_AV;
- }
- }
-
- //duration is set to 20 ms worth of stereo data at 48Khz
+ //duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
@@ -3377,29 +3557,61 @@
return fragment_size;
}
+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")) ||
+ !strncmp(snd_card_name, "msm8976-tasha-skun-snd-card",
+ sizeof("msm8976-tasha-skun-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;
- const char *snd_card_name = mixer_get_name(adev->mixer);
+ 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);
- 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")) ||
- !strncmp(snd_card_name, "msm8976-tasha-snd-card",
- sizeof("msm8976-tasha-snd-card"))) {
- mixer_ctl_name = "SLIM_0_RX Format";
- }
- else
- mixer_ctl_name = "MI2S_RX Format";
+ const char *snd_card_name = mixer_get_name(adev->mixer);
+ 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;
}
@@ -3407,10 +3619,12 @@
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);
}
/*
@@ -3422,9 +3636,12 @@
* 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) {
@@ -3433,10 +3650,12 @@
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:
@@ -3451,32 +3670,46 @@
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
@@ -3497,52 +3730,50 @@
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;
+ }
}
-
- //check if mulitchannel clip needs to be down sampled to 48k
- property_get("audio.playback.mch.downsample",value,"");
- if (!strncmp("true", value, sizeof("true"))) {
- out = usecase->stream.out;
- if ((popcount(out->channel_mask) > 2) &&
- (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
- !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- /* update out sample rate to reflect current backend sample rate */
- out->sample_rate = sample_rate;
- ALOGD("%s: MCH session defaulting sample rate to %d",
- __func__, 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;
@@ -3553,20 +3784,27 @@
return backend_change;
}
-bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase)
+bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
+ 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;
}
@@ -3819,10 +4057,15 @@
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);
- strlcat(mixer_ctl_name, device_num, sizeof(device_num));
+ 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);
@@ -4238,3 +4481,16 @@
done:
return ret;
}
+
+int platform_get_wsa_mode(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ char *snd_card_name;
+
+ snd_card_name = mixer_get_name(my_data->adev->mixer);
+ if ((!strcmp(snd_card_name, "msm8952-skum-snd-card")) ||
+ (!strcmp(snd_card_name, "msm8952-snd-card-mtp")))
+ return 1;
+ else
+ return 0;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index d564839..9f0015a 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 | \
@@ -60,6 +64,7 @@
SND_DEVICE_OUT_SPEAKER_REVERSE,
SND_DEVICE_OUT_SPEAKER_WSA,
SND_DEVICE_OUT_HEADPHONES,
+ SND_DEVICE_OUT_HEADPHONES_44_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2,
@@ -162,6 +167,16 @@
};
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#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
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index bceaf96..038effa 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -676,7 +676,7 @@
goto exit;
}
- if (voice_is_in_call(adev)) {
+ if (mode == AUDIO_MODE_IN_CALL) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
if (adev->voice.tty_mode == TTY_MODE_FULL)
@@ -768,7 +768,7 @@
ALOGV("%s: enter: out_device(%#x) in_device(%#x)",
__func__, out_device, in_device);
- if ((out_device != AUDIO_DEVICE_NONE) && voice_is_in_call(adev)) {
+ if ((out_device != AUDIO_DEVICE_NONE) && (mode == AUDIO_MODE_IN_CALL)) {
if (adev->voice.tty_mode != TTY_MODE_OFF) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
@@ -1180,3 +1180,8 @@
{
return -ENOSYS;
}
+
+int platform_get_wsa_mode(void *platform)
+{
+ return 0;
+}
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 461c9a2..8c1a768 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -65,17 +65,14 @@
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
/* Used in calculating fragment size for pcm offload */
-#define PCM_OFFLOAD_BUFFER_DURATION_FOR_AV 1000 /* 1 sec */
-#define PCM_OFFLOAD_BUFFER_DURATION_FOR_AV_STREAMING 80 /* 80 millisecs */
-#define PCM_OFFLOAD_BUFFER_DURATION_FOR_SMALL_BUFFERS 20 /* 20 millisecs */
-#define PCM_OFFLOAD_BUFFER_DURATION_MAX 1200 /* 1200 millisecs */
+#define PCM_OFFLOAD_BUFFER_DURATION 40 /* 40 millisecs */
/* MAX PCM fragment size cannot be increased further due
* to flinger's cblk size of 1mb,and it has to be a multiple of
* 24 - lcm of channels supported by DSP
*/
#define MAX_PCM_OFFLOAD_FRAGMENT_SIZE (240 * 1024)
-#define MIN_PCM_OFFLOAD_FRAGMENT_SIZE (4 * 1024)
+#define MIN_PCM_OFFLOAD_FRAGMENT_SIZE 512
/*
* Offload buffer size for compress passthrough
@@ -171,6 +168,20 @@
typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
+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;
@@ -187,6 +198,7 @@
bool hd_voice;
bool ec_ref_enabled;
bool is_i2s_ext_modem;
+ bool is_acdb_initialized;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -203,6 +215,7 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
+ codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
};
static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -287,6 +300,7 @@
[SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = "speaker-ext-2",
[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",
@@ -384,6 +398,7 @@
[SND_DEVICE_OUT_SPEAKER_EXTERNAL_2] = 130,
[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] = 130,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = 130,
@@ -475,7 +490,7 @@
#define TO_NAME_INDEX(X) #X, X
-/* Used to get index from parsed sting */
+/* Used to get index from parsed string */
static struct name_to_index snd_device_name_index[SND_DEVICE_MAX] = {
{TO_NAME_INDEX(SND_DEVICE_OUT_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER)},
@@ -483,6 +498,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_EXTERNAL_2)},
{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)},
@@ -878,7 +894,7 @@
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.
@@ -901,6 +917,8 @@
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] = strdup("headphones");
+ backend_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
}
void get_cvd_version(char *cvd_version, struct audio_device *adev)
@@ -1016,6 +1034,37 @@
ALOGE("%s: Could not send anc cal", __FUNCTION__);
}
+int platform_acdb_init(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ char *cvd_version = NULL;
+ int key = 0;
+ const char *snd_card_name;
+ int result;
+ char value[PROPERTY_VALUE_MAX];
+ cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
+ if (!cvd_version)
+ ALOGE("Failed to allocate cvd version");
+ else
+ get_cvd_version(cvd_version, my_data->adev);
+
+ property_get("audio.ds1.metainfo.key",value,"0");
+ key = atoi(value);
+ snd_card_name = mixer_get_name(my_data->adev->mixer);
+ result = my_data->acdb_init(snd_card_name, cvd_version, key);
+ if (cvd_version)
+ free(cvd_version);
+ if (!result) {
+ my_data->is_acdb_initialized = true;
+ ALOGD("ACDB initialized");
+ audio_hwdep_send_cal(my_data);
+ } else {
+ my_data->is_acdb_initialized = false;
+ ALOGD("ACDB initialization failed");
+ }
+ return result;
+}
+
void *platform_init(struct audio_device *adev)
{
char platform[PROPERTY_VALUE_MAX];
@@ -1030,6 +1079,7 @@
char mixer_xml_file[MIXER_PATH_MAX_LENGTH]= {0};
const char *mixer_ctl_name = "Set HPX ActiveBe";
struct mixer_ctl *ctl = NULL;
+ int idx;
my_data = calloc(1, sizeof(struct platform_data));
@@ -1182,8 +1232,6 @@
my_data->fluence_mode = FLUENCE_BROADSIDE;
}
}
- property_get("audio.ds1.metainfo.key",value,"0");
- key = atoi(value);
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
@@ -1241,15 +1289,7 @@
goto acdb_init_fail;
}
- cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
- if (!cvd_version)
- ALOGE("failed to allocate cvd_version");
- else
- get_cvd_version(cvd_version, adev);
-
- my_data->acdb_init(snd_card_name, cvd_version, key);
- if (cvd_version)
- free(cvd_version);
+ platform_acdb_init(my_data);
}
/* Configure active back end for HPX*/
@@ -1300,6 +1340,24 @@
/* init audio device arbitration */
audio_extn_dev_arbi_init();
+ /* 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;
}
@@ -1338,6 +1396,13 @@
audio_extn_dap_hal_deinit();
}
+int platform_is_acdb_initialized(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ ALOGD("%s: acdb initialized %d\n", __func__, my_data->is_acdb_initialized);
+ return my_data->is_acdb_initialized;
+}
+
const char *platform_get_snd_device_name(snd_device_t snd_device)
{
if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX)
@@ -1565,6 +1630,111 @@
return backend_bit_width_table[snd_device];
}
+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_native_support()
+{
+ 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") == 0)
+ 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,
int app_type, int sample_rate)
{
@@ -1575,7 +1745,7 @@
if (usecase->type == PCM_PLAYBACK)
snd_device = platform_get_output_snd_device(adev->platform,
- usecase->stream.out->devices);
+ usecase->stream.out);
else if ((usecase->type == PCM_HFP_CALL) || (usecase->type == PCM_CAPTURE))
snd_device = platform_get_input_snd_device(adev->platform,
adev->primary_output->devices);
@@ -1856,12 +2026,14 @@
return ret;
}
-snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
+snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)
{
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
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;
audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
@@ -1874,7 +2046,7 @@
goto exit;
}
- if (popcount(devices) == 2) {
+ if (popcount(devices) == 2 && !voice_is_in_call(adev)) {
if (devices == (AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
AUDIO_DEVICE_OUT_SPEAKER)) {
if (my_data->external_spk_1)
@@ -1913,7 +2085,7 @@
goto exit;
}
- if (voice_is_in_call(adev) ||
+ if ((mode == AUDIO_MODE_IN_CALL) ||
voice_extn_compress_voip_is_active(adev)) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
@@ -1974,8 +2146,11 @@
snd_device = SND_DEVICE_OUT_ANC_FB_HEADSET;
else
snd_device = SND_DEVICE_OUT_ANC_HEADSET;
- } else
- snd_device = SND_DEVICE_OUT_HEADPHONES;
+ } else if (platform_get_native_support() &&
+ OUTPUT_SAMPLING_RATE_44100 == sample_rate)
+ snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+ else
+ snd_device = SND_DEVICE_OUT_HEADPHONES;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
if (my_data->external_spk_1)
snd_device = SND_DEVICE_OUT_SPEAKER_EXTERNAL_1;
@@ -2033,8 +2208,8 @@
ALOGV("%s: enter: out_device(%#x) in_device(%#x)",
__func__, out_device, in_device);
if (my_data->external_mic) {
- if ((out_device != AUDIO_DEVICE_NONE && voice_is_in_call(adev)) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev)) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_EARPIECE ||
out_device & AUDIO_DEVICE_OUT_SPEAKER )
@@ -2048,7 +2223,7 @@
if (snd_device != AUDIO_DEVICE_NONE)
goto exit;
- if ((out_device != AUDIO_DEVICE_NONE) && ((voice_is_in_call(adev)) ||
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
!voice_extn_compress_voip_is_active(adev)) {
@@ -2528,6 +2703,7 @@
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;
acdb_audio_cal_cfg_t cal={0};
uint8_t *dptr = NULL;
int32_t dlen;
@@ -2564,7 +2740,8 @@
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);
@@ -2680,7 +2857,7 @@
/* handle audio calibration parameters */
set_audiocal(platform, parms, value, len);
-
+ native_audio_set_params(platform, parms, value, len);
done:
ALOGV("%s: exit with code(%d)", __func__, ret);
if(kv_pairs != NULL)
@@ -2792,6 +2969,7 @@
static void get_audiocal(void *platform, void *keys, void *pReply) {
struct platform_data *my_data = (struct platform_data *)platform;
+ struct stream_out out;
struct str_parms *query = (struct str_parms *)keys;
struct str_parms *reply=(struct str_parms *)pReply;
acdb_audio_cal_cfg_t cal={0};
@@ -2822,7 +3000,8 @@
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) {
@@ -2921,6 +3100,7 @@
/* Handle audio calibration keys */
get_audiocal(platform, query, reply);
+ native_audio_get_params(query, reply, value, sizeof(value));
done:
kv_pairs = str_parms_to_str(reply);
@@ -3026,25 +3206,13 @@
{
uint32_t fragment_size = 0;
uint32_t bits_per_sample = 16;
- uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_SMALL_BUFFERS;
+ uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
bits_per_sample = 32;
}
- if (info->use_small_bufs) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_SMALL_BUFFERS;
- } else {
- if (!info->has_video) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_MAX;
- } else if (info->has_video && info->is_streaming) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_AV_STREAMING;
- } else if (info->has_video) {
- pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION_FOR_AV;
- }
- }
-
- //duration is set to 20 ms worth of stereo data at 48Khz
+ //duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
@@ -3070,29 +3238,39 @@
}
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;
- if (bit_width != adev->cur_codec_backend_bit_width) {
- const char * mixer_ctl_name = "SLIM_0_RX Format";
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+
+ backend_idx = platform_get_backend_index(snd_device);
+ ALOGV("%s bit width: %d, sample rate: %d backend_idx - %d",
+ __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);
+ 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;
}
if (bit_width == 24) {
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ 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);
}
/*
@@ -3103,10 +3281,9 @@
* 24 bit playback - 192khz for sample rate range of 96khz to 192 khz
* 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[backend_idx].sample_rate) {
char *rate_str = NULL;
- const char * mixer_ctl_name = "SLIM_0_RX SampleRate";
struct mixer_ctl *ctl;
switch (sample_rate) {
@@ -3115,10 +3292,12 @@
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:
@@ -3133,33 +3312,46 @@
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;
+ unsigned int bit_width;
+ unsigned int sample_rate;
char value[PROPERTY_VALUE_MAX] = {0};
+ 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
if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
@@ -3179,51 +3371,63 @@
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;
}
}
}
}
-
- // 16 bit playback on speakers is allowed through 48 khz backend only
- if (16 == bit_width) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- }
-
- //check if mulitchannel clip needs to be down sampled to 48k
- property_get("audio.playback.mch.downsample",value,"");
- if (!strncmp("true", value, sizeof("true"))) {
- out = usecase->stream.out;
- if ((popcount(out->channel_mask) > 2) &&
- (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
- !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
- /* update out sample rate to reflect current backend 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: MCH session defaulting sample rate to %d",
- __func__, sample_rate);
+ ALOGD("%s: resetting sample_rate back to default, "
+ "backend_idx: %d", __func__, backend_idx);
+ }
+
+ //check if mulitchannel clip needs to be down sampled to 48k
+ property_get("audio.playback.mch.downsample",value,"");
+ if (!strncmp("true", value, sizeof("true"))) {
+ out = usecase->stream.out;
+ if ((popcount(out->channel_mask) > 2) &&
+ (out->sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
+ !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
+ /* update out sample rate to reflect current backend sample rate */
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s: MCH session defaulting sample rate to %d",
+ __func__, 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;
@@ -3234,20 +3438,27 @@
return backend_change;
}
-bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase)
+bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
+ 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;
}
@@ -3883,3 +4094,8 @@
done:
return ret;
}
+
+int platform_get_wsa_mode (void *platform)
+{
+ return 0;
+}
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 41e2c55..4b90f8b 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -35,7 +35,11 @@
/*
* 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 | \
@@ -57,6 +61,7 @@
SND_DEVICE_OUT_SPEAKER_EXTERNAL_2,
SND_DEVICE_OUT_SPEAKER_REVERSE,
SND_DEVICE_OUT_HEADPHONES,
+ SND_DEVICE_OUT_HEADPHONES_44_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2,
@@ -151,7 +156,16 @@
};
-#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#define OUTPUT_SAMPLING_RATE_44100 44100
+
+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
diff --git a/hal/platform_api.h b/hal/platform_api.h
index b2a0ccc..3a6dc02 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -26,6 +26,8 @@
void *platform_init(struct audio_device *adev);
void platform_deinit(void *platform);
+int platform_is_acdb_initialized(void *platform);
+int platform_acdb_init(void *platform);
const char *platform_get_snd_device_name(snd_device_t snd_device);
int platform_get_snd_device_name_extn(void *platform, snd_device_t snd_device,
char *device_name);
@@ -38,6 +40,9 @@
int platform_get_snd_device_acdb_id(snd_device_t snd_device);
int platform_set_snd_device_bit_width(snd_device_t snd_device, unsigned int bit_width);
int platform_get_snd_device_bit_width(snd_device_t snd_device);
+int platform_set_native_support(bool codec_support);
+int platform_get_native_support();
+int platform_get_backend_index(snd_device_t snd_device);
int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
int app_type, int sample_rate);
int platform_get_default_app_type(void *platform);
@@ -57,7 +62,7 @@
int platform_set_mic_mute(void *platform, bool state);
int platform_get_sample_rate(void *platform, uint32_t *rate);
int platform_set_device_mute(void *platform, bool state, char *dir);
-snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices);
+snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out);
snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device);
int platform_set_hdmi_channels(void *platform, int channel_count);
int platform_edid_get_max_channels(void *platform);
@@ -92,7 +97,8 @@
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, struct audio_usecase *usecase);
+bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
+ struct audio_usecase *usecase, snd_device_t snd_device);
int platform_get_usecase_index(const char * usecase);
int platform_set_usecase_pcm_id(audio_usecase_t usecase, int32_t type, int32_t pcm_id);
void platform_set_echo_reference(void *platform, bool enable);
@@ -116,4 +122,5 @@
const char *spkr_1_tz_name, const char *spkr_2_tz_name);
const char *platform_get_spkr_1_tz_name(snd_device_t snd_device);
const char *platform_get_spkr_2_tz_name(snd_device_t snd_device);
+int platform_get_wsa_mode(void *platform);
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 6a35be5..50e19c6 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -44,6 +44,7 @@
ROOT,
ACDB,
BITWIDTH,
+ NATIVESUPPORT,
PCM_ID,
BACKEND_NAME,
INTERFACE_NAME,
@@ -54,6 +55,7 @@
static void process_acdb_id(const XML_Char **attr);
static void process_bit_width(const XML_Char **attr);
+static void process_native_support(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);
@@ -64,6 +66,7 @@
[ROOT] = process_root,
[ACDB] = process_acdb_id,
[BITWIDTH] = process_bit_width,
+ [NATIVESUPPORT] = process_native_support,
[PCM_ID] = process_pcm_id,
[BACKEND_NAME] = process_backend_name,
[INTERFACE_NAME] = process_interface_name,
@@ -281,7 +284,30 @@
(char *)attr[5]);
if (ret < 0) {
ALOGE("%s: Audio Interface not set!", __func__);
+ goto done;
+ }
+done:
+ return;
+}
+
+static void process_native_support(const XML_Char **attr)
+{
+ int index;
+
+ if (strcmp(attr[0], "name") != 0) {
+ ALOGE("%s: 'name' not found, no NATIVE_AUDIO_44.1 set!", __func__);
+ goto done;
+ }
+
+ if (strcmp(attr[2], "codec_support") != 0) {
+ ALOGE("%s: NATIVE_AUDIO_44.1 in platform info xml has no codec_support set!",
+ __func__);
+ goto done;
+ }
+
+ if (platform_set_native_support(atoi((char *)attr[3])) < 0) {
+ ALOGE("%s: NATIVE_AUDIO_44 was not set!", __func__);
goto done;
}
@@ -344,6 +370,8 @@
section = INTERFACE_NAME;
} else if (strcmp(tag_name, "tz_names") == 0) {
section = TZ_NAME;
+ } else if (strcmp(tag_name, "native_configs") == 0) {
+ section = NATIVESUPPORT;
} else if (strcmp(tag_name, "device") == 0) {
if ((section != ACDB) && (section != BACKEND_NAME) && (section != BITWIDTH) &&
(section != INTERFACE_NAME) && (section != TZ_NAME)) {
@@ -362,6 +390,14 @@
section_process_fn fn = section_table[PCM_ID];
fn(attr);
+ } else if (strcmp(tag_name, "feature") == 0) {
+ if (section != NATIVESUPPORT) {
+ ALOGE("usecase tag only supported with NATIVESUPPORT section");
+ return;
+ }
+
+ section_process_fn fn = section_table[NATIVESUPPORT];
+ fn(attr);
}
return;
@@ -379,6 +415,8 @@
section = ROOT;
} else if (strcmp(tag_name, "interface_names") == 0) {
section = ROOT;
+ } else if (strcmp(tag_name, "native_configs") == 0) {
+ section = ROOT;
}
}
diff --git a/hal/voice.c b/hal/voice.c
index c00df1a..527856f 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -56,6 +56,59 @@
return session;
}
+static bool voice_is_sidetone_device(snd_device_t out_device,
+ char *mixer_path)
+{
+ bool is_sidetone_dev;
+
+ switch (out_device) {
+ case SND_DEVICE_OUT_VOICE_HANDSET:
+ is_sidetone_dev = true;
+ strlcpy(mixer_path, "sidetone-handset", MIXER_PATH_MAX_LENGTH);
+ break;
+ case SND_DEVICE_OUT_VOICE_HEADPHONES:
+ case SND_DEVICE_OUT_VOICE_ANC_HEADSET:
+ case SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET:
+ is_sidetone_dev = true;
+ strlcpy(mixer_path, "sidetone-headphones", MIXER_PATH_MAX_LENGTH);
+ break;
+ default:
+ is_sidetone_dev = false;
+ break;
+ }
+
+ return is_sidetone_dev;
+}
+
+void voice_set_sidetone(struct audio_device *adev,
+ snd_device_t out_snd_device, bool enable)
+{
+ char mixer_path[MIXER_PATH_MAX_LENGTH];
+ bool is_sidetone_dev;
+
+ ALOGD("%s: %s, out_snd_device: %d\n",
+ __func__, (enable ? "enable" : "disable"),
+ out_snd_device);
+
+ is_sidetone_dev = voice_is_sidetone_device(out_snd_device, mixer_path);
+
+ if (!is_sidetone_dev) {
+ ALOGD("%s: device %d does not support sidetone\n",
+ __func__, out_snd_device);
+ return;
+ }
+
+ ALOGD("%s: sidetone out device = %s\n",
+ __func__, mixer_path);
+
+ if (enable)
+ audio_route_apply_and_update_path(adev->audio_route, mixer_path);
+ else
+ audio_route_reset_and_update_path(adev->audio_route, mixer_path);
+
+ return;
+}
+
int voice_stop_usecase(struct audio_device *adev, audio_usecase_t usecase_id)
{
int i, ret = 0;
@@ -70,10 +123,21 @@
return -EINVAL;
}
+ uc_info = get_usecase_from_list(adev, usecase_id);
+ if (uc_info == NULL) {
+ ALOGE("%s: Could not find the usecase (%d) in the list",
+ __func__, usecase_id);
+ return -EINVAL;
+ }
+
session->state.current = CALL_INACTIVE;
if (adev->mode == AUDIO_MODE_NORMAL)
adev->voice.is_in_call = false;
+ /* Disable sidetone only when no calls are active */
+ if (!voice_is_call_state_active(adev))
+ voice_set_sidetone(adev, uc_info->out_snd_device, false);
+
ret = platform_stop_voice_call(adev->platform, session->vsid);
/* 1. Close the PCM devices */
@@ -86,13 +150,6 @@
session->pcm_tx = NULL;
}
- uc_info = get_usecase_from_list(adev, usecase_id);
- if (uc_info == NULL) {
- ALOGE("%s: Could not find the usecase (%d) in the list",
- __func__, usecase_id);
- return -EINVAL;
- }
-
/* 2. Get and set stream specific mixer controls */
disable_audio_route(adev, uc_info);
@@ -160,6 +217,17 @@
voice_set_mic_mute(adev, adev->voice.mic_mute);
+ ALOGV("%s: Opening PCM capture device card_id(%d) device_id(%d)",
+ __func__, adev->snd_card, pcm_dev_tx_id);
+ session->pcm_tx = pcm_open(adev->snd_card,
+ pcm_dev_tx_id,
+ PCM_IN, &voice_config);
+ if (session->pcm_tx && !pcm_is_ready(session->pcm_tx)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(session->pcm_tx));
+ ret = -EIO;
+ goto error_start_voice;
+ }
+
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,
@@ -171,18 +239,12 @@
goto error_start_voice;
}
- ALOGV("%s: Opening PCM capture device card_id(%d) device_id(%d)",
- __func__, adev->snd_card, pcm_dev_tx_id);
- session->pcm_tx = pcm_open(adev->snd_card,
- pcm_dev_tx_id,
- PCM_IN, &voice_config);
- if (session->pcm_tx && !pcm_is_ready(session->pcm_tx)) {
- ALOGE("%s: %s", __func__, pcm_get_error(session->pcm_tx));
- ret = -EIO;
- goto error_start_voice;
- }
- pcm_start(session->pcm_rx);
pcm_start(session->pcm_tx);
+ pcm_start(session->pcm_rx);
+
+ /* Enable sidetone only when no calls are already active */
+ if (!voice_is_call_state_active(adev))
+ voice_set_sidetone(adev, uc_info->out_snd_device, true);
voice_set_volume(adev, adev->voice.volume);
diff --git a/hal/voice.h b/hal/voice.h
index 5a9cce1..139a8c8 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -95,4 +95,8 @@
struct stream_in *in);
void voice_update_devices_for_all_voice_usecases(struct audio_device *adev);
snd_device_t voice_get_incall_rec_snd_device(snd_device_t in_snd_device);
+void voice_set_sidetone(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable);
+bool voice_is_call_state_active(struct audio_device *adev);
#endif //VOICE_H
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/bundle.c b/post_proc/bundle.c
index c09d2ad..b33f2d1 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -304,9 +304,6 @@
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,
@@ -315,6 +312,9 @@
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