hal: Changes to support true 44.1 Native playback
- add changes to support single BE mode on top of the
Double BE mode which is supported.
- support reading of version or mode from XML.
- add logic to switch between single and double BE
based on the mode in XML.
- define a new device for true native.
- add logic to read tasha codec version and allow
true native only for 2.0 version of codec
- enchance log msgs for better debugging.
Change-Id: I970487f16af32144ec26f967e8bfcb32d44d67a4
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 8c17895..8ad9922 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -198,11 +198,7 @@
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};
+static native_audio_prop na_props = {0, 0, 0};
typedef int (*acdb_send_gain_dep_cal_t)(int, int, int, int, int);
struct platform_data {
@@ -246,6 +242,7 @@
bool edid_valid;
char ec_ref_mixer_path[64];
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
+ char codec_version[CODEC_VERSION_MAX_LENGTH];
};
static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -1273,7 +1270,7 @@
char value[PROPERTY_VALUE_MAX];
struct platform_data *my_data = NULL;
int retry_num = 0, snd_card_num = 0, key = 0;
- const char *snd_card_name = NULL;
+ const char *snd_card_name = NULL, *snd_card_name_t = NULL;
char *cvd_version = NULL;
char *snd_internal_name = NULL;
char *tmp = NULL;
@@ -1310,7 +1307,7 @@
free(my_data);
return NULL;
}
- ALOGV("%s: snd_card_name: %s", __func__, snd_card_name);
+ ALOGD("%s: snd_card_name: %s", __func__, snd_card_name);
my_data->hw_info = hw_info_init(snd_card_name);
if (!my_data->hw_info) {
@@ -1334,8 +1331,9 @@
* done to preserve backward compatibility but not mandatory as
* long as the mixer files are named as per above assumption.
*/
+ snd_card_name_t = strdup(snd_card_name);
+ snd_internal_name = strtok_r(snd_card_name_t, "-", &tmp);
- snd_internal_name = strtok_r(snd_card_name, "-", &tmp);
if (snd_internal_name != NULL)
snd_internal_name = strtok_r(NULL, "-", &tmp);
@@ -1372,6 +1370,7 @@
__func__);
free(my_data);
free(snd_card_name);
+ free(snd_card_name_t);
return NULL;
}
adev->snd_card = snd_card_num;
@@ -1387,6 +1386,8 @@
free(my_data);
if (snd_card_name)
free(snd_card_name);
+ if (snd_card_name_t)
+ free(snd_card_name_t);
return NULL;
}
@@ -1584,8 +1585,29 @@
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_5_RX SampleRate");
+ ret = audio_extn_utils_get_codec_version(snd_card_name,
+ my_data->adev->snd_card,
+ my_data->codec_version);
+
+ if (NATIVE_AUDIO_MODE_INVALID != platform_get_native_support()) {
+ /*
+ * Native playback is enabled from the UI.
+ */
+ if(strstr(snd_card_name, "tasha")) {
+ if (strstr(my_data->codec_version, "WCD9335_1_0") ||
+ strstr(my_data->codec_version, "WCD9335_1_1")) {
+ ALOGD("%s:napb: TASHA 1.0 or 1.1 only SRC mode is supported",
+ __func__);
+ platform_set_native_support(NATIVE_AUDIO_MODE_SRC);
+ }
+ } else {
+ platform_set_native_support(NATIVE_AUDIO_MODE_INVALID);
+ }
+ }
+
my_data->edid_info = NULL;
free(snd_card_name);
+ free(snd_card_name_t);
return my_data;
}
@@ -1870,26 +1892,32 @@
return backend_bit_width_table[snd_device];
}
-int platform_set_native_support(bool codec_support)
+int platform_set_native_support(int na_mode)
{
- int ret = 0;
- na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
- = codec_support;
- ALOGD("%s: na_props.platform_na_prop_enabled: %d", __func__,
- na_props.platform_na_prop_enabled);
- return ret;
+ if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode) {
+ na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled = true;
+ na_props.na_mode = na_mode;
+ ALOGD("%s:napb: native audio playback enabled in (%s) mode", __func__,
+ ((na_mode == NATIVE_AUDIO_MODE_SRC)?"SRC mode":"True 44.1 mode"));
+ }
+ else {
+ na_props.platform_na_prop_enabled = false;
+ na_props.na_mode = NATIVE_AUDIO_MODE_INVALID;
+ ALOGD("%s:napb: native audio playback disabled", __func__);
+ }
+
+ 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;
+ int ret = NATIVE_AUDIO_MODE_INVALID;
+ if (na_props.platform_na_prop_enabled &&
+ na_props.ui_na_prop_enabled) {
+ ret = na_props.na_mode;
}
- ALOGV("%s: na_props.ui_na_prop_enabled: %d", __func__,
- na_props.ui_na_prop_enabled);
+ ALOGV("%s:napb: ui Prop enabled(%d) version(%d)", __func__,
+ na_props.ui_na_prop_enabled, na_props.na_mode);
return ret;
}
@@ -1904,13 +1932,13 @@
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);
+ ALOGV("%s:napb: 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);
+ ALOGV("%s:napb: native audio not supported: %d", __func__,
+ na_props.platform_na_prop_enabled);
}
}
}
@@ -1921,6 +1949,25 @@
int ret = 0;
struct audio_usecase *usecase;
struct listnode *node;
+ int mode = NATIVE_AUDIO_MODE_INVALID;
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO_MODE,
+ value, len);
+ if (ret >= 0) {
+ if (value && !strncmp(value, "src", sizeof("src")))
+ mode = NATIVE_AUDIO_MODE_SRC;
+ else if (value && !strncmp(value, "true", sizeof("true")))
+ mode = NATIVE_AUDIO_MODE_TRUE_44_1;
+ else {
+ mode = NATIVE_AUDIO_MODE_INVALID;
+ ALOGE("%s:napb:native_audio_mode in platform info xml,invalid mode string",
+ __func__);
+ }
+ ALOGD("%s:napb updating mode (%d) from XML",__func__, mode);
+ platform_set_native_support(mode);
+ }
+
+
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
value, len);
@@ -1928,12 +1975,12 @@
if (na_props.platform_na_prop_enabled) {
if (!strncmp("true", value, sizeof("true"))) {
na_props.ui_na_prop_enabled = true;
- ALOGD("%s: native audio feature enabled from UI",__func__);
- }
- else {
+ ALOGD("%s:napb: native audio feature enabled from UI",
+ __func__);
+ } else {
na_props.ui_na_prop_enabled = false;
- ALOGD("%s: native audio feature disabled from UI",__func__);
-
+ ALOGD("%s:napb: native audio feature disabled from UI",
+ __func__);
}
str_parms_del(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO);
@@ -1949,21 +1996,46 @@
(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) {
- ALOGD("%s: triggering dynamic device switch for usecase(%d: %s)"
+ ALOGD("%s:napb: triggering dynamic device switch for usecase(%d: %s)"
" stream(%p), device(%d)", __func__, usecase->id,
use_case_table[usecase->id], usecase->stream,
usecase->stream.out->devices);
select_devices(platform->adev, usecase->id);
}
}
- } else {
- ALOGD("%s: native audio not supported: %d", __func__,
- na_props.platform_na_prop_enabled);
- }
+ } else
+ ALOGD("%s:napb: native audio cannot be enabled from UI",
+ __func__);
}
return ret;
}
+int check_hdset_combo_device(struct audio_device *adev, snd_device_t snd_device)
+{
+ int ret = false;
+ struct listnode *node;
+ int i =0;
+
+ if (SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES == snd_device)
+ ret = true;
+ else {
+ list_for_each(node, &adev->usecase_list) {
+ struct audio_usecase *uc;
+ uc = node_to_item(node, struct audio_usecase, list);
+ ALOGD("%s: (%d) use case %s snd device %s",
+ __func__, i++, use_case_table[uc->id],
+ platform_get_snd_device_name(uc->out_snd_device));
+
+ if (SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES == uc->out_snd_device)
+ ret = true;
+ }
+ }
+ ALOGV("%s:napb: (%s) snd_device (%s)",
+ __func__, (ret == false ? "false":"true"),
+ platform_get_snd_device_name(snd_device));
+
+ return ret;
+}
int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -1975,10 +2047,10 @@
else
port = DEFAULT_CODEC_BACKEND;
} else {
- ALOGV("%s: Invalid device - %d ", __func__, snd_device);
+ ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
}
- ALOGV("%s: backend port - %d", __func__, port);
+ ALOGV("%s:napb: backend port - %d", __func__, port);
return port;
}
@@ -2291,6 +2363,7 @@
snd_device_t snd_device = SND_DEVICE_NONE;
audio_devices_t devices = out->devices;
unsigned int sample_rate = out->sample_rate;
+ int na_mode = platform_get_native_support();
audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
@@ -2412,11 +2485,11 @@
snd_device = SND_DEVICE_OUT_ANC_FB_HEADSET;
else
snd_device = SND_DEVICE_OUT_ANC_HEADSET;
- } else if (platform_get_native_support() &&
- OUTPUT_SAMPLING_RATE_44100 == sample_rate)
+ } else if (NATIVE_AUDIO_MODE_SRC == na_mode &&
+ OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
- else
- snd_device = SND_DEVICE_OUT_HEADPHONES;
+ } else
+ snd_device = SND_DEVICE_OUT_HEADPHONES;
} else if (devices & AUDIO_DEVICE_OUT_LINE) {
snd_device = SND_DEVICE_OUT_LINE;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
@@ -3591,6 +3664,9 @@
return OFFLOAD_USE_SMALL_BUFFER;
}
+/*
+ * configures afe with bit width and Sample Rate
+ */
int platform_set_codec_backend_cfg(struct audio_device* adev,
snd_device_t snd_device,
unsigned int bit_width, unsigned int sample_rate)
@@ -3600,8 +3676,10 @@
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);
+
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
+ ", backend_idx %d device (%s)", __func__, bit_width, sample_rate, backend_idx,
+ platform_get_snd_device_name(snd_device));
if (bit_width !=
my_data->current_backend_cfg[backend_idx].bit_width) {
@@ -3610,8 +3688,9 @@
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__, my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
return -EINVAL;
}
@@ -3619,12 +3698,10 @@
mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
- if (backend_idx != HEADPHONE_44_1_BACKEND)
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
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);
+ ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
}
/*
@@ -3669,13 +3746,14 @@
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__, my_data->current_backend_cfg[backend_idx].samplerate_mixer_ctl);
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__,
+ my_data->current_backend_cfg[backend_idx].samplerate_mixer_ctl);
return -EINVAL;
}
- ALOGD("%s: %s set to %s", __func__,
- my_data->current_backend_cfg[backend_idx].samplerate_mixer_ctl, rate_str);
+ ALOGD("%s:becf: afe: %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);
my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
}
@@ -3683,6 +3761,10 @@
return ret;
}
+/*
+ * goes through all the current usecases and picks the highest
+ * bitwidth & samplerate
+ */
bool platform_check_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase* usecase,
snd_device_t snd_device,
@@ -3697,21 +3779,25 @@
int backend_idx = DEFAULT_CODEC_BACKEND;
int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
+ int na_mode = platform_get_native_support();
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);
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
+ ", backend_idx %d usecase = %d device (%s)", __func__, bit_width,
+ sample_rate, backend_idx, usecase->id,
+ platform_get_snd_device_name(snd_device));
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
// force routing is not required here, caller will do it anyway
if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
backend_idx == DEFAULT_CODEC_BACKEND) {
- ALOGW("%s:Use default bw and sr for voice/voip calls ",__func__);
+ ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
+ __func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
} else {
@@ -3722,22 +3808,24 @@
* current backend sample rate and/or bit width, then, we set the
* backend re-configuration flag.
*
- * Exception: 16 bit playbacks is allowed through 16 bit/48 khz backend only
+ * Exception: 16 bit playbacks is allowed through 16 bit/48/44.1 khz backend only
*/
+ int i =0;
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 &&
- usecase != curr_usecase) {
- struct stream_out *out =
- (struct stream_out*) curr_usecase->stream.out;
- usecase_backend_idx = platform_get_backend_index(curr_usecase->out_snd_device);
+ struct audio_usecase *uc;
+ uc = node_to_item(node, struct audio_usecase, list);
+ struct stream_out *out = (struct stream_out*) uc->stream.out;
+ if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+ usecase_backend_idx =
+ platform_get_backend_index(uc->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);
+ ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
+ "(%d) device %s", __func__, i++, use_case_table[uc->id],
+ uc->id, out->sample_rate,
+ out->bit_width,
+ platform_get_snd_device_name(uc->out_snd_device));
+
+ if (usecase_backend_idx == backend_idx) {
if (bit_width < out->bit_width)
bit_width = out->bit_width;
if (sample_rate < out->sample_rate)
@@ -3749,24 +3837,52 @@
}
}
- 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) {
+ if (audio_is_true_native_stream_active(adev)) {
+ if (check_hdset_combo_device(adev, snd_device)) {
+ /*
+ * In true native mode Tasha has a limitation that one port at 44.1 khz
+ * cannot drive both spkr and hdset, to simiplify the solution lets
+ * move the AFE to 48khzwhen a ring tone selects combo device.
+ */
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s: resetting sample_rate back to default, "
- "backend_idx: %d", __func__, backend_idx);
- }
- // 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;
+ bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
+ __func__);
+ } else {
+ /*
+ * in single BE mode, if native audio playback
+ * is active then it will take priority
+ */
+ sample_rate = OUTPUT_SAMPLING_RATE_44100;
+ ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
+ __func__);
}
}
- ALOGI("%s Codec selected backend: %d updated bit width: %d and sample rate: %d",
- __func__, backend_idx, bit_width, sample_rate);
+
+ /*
+ * 24 bit playback on speakers is allowed through 48
+ * khz backend only
+ */
+ 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;
+ ALOGD("%s:becf: afe: 24 bit playback on speakers"
+ "Configure afe to default Sample Rate(48k)", __func__);
+ }
+
+ /*
+ * native playback is not enabled.Configure afe to default Sample Rate(48k)
+ */
+ if (NATIVE_AUDIO_MODE_INVALID == na_mode &&
+ OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+ __func__);
+ }
+
+ ALOGI("%s:becf: afe: 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 != my_data->current_backend_cfg[backend_idx].bit_width) ||
@@ -3774,8 +3890,8 @@
*new_bit_width = bit_width;
*new_sample_rate = sample_rate;
backend_change = true;
- ALOGI("%s Codec backend needs to be updated. new bit width: %d new sample rate: %d",
- __func__, *new_bit_width, *new_sample_rate);
+ ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
+ __func__, *new_bit_width, *new_sample_rate);
}
return backend_change;
@@ -3789,15 +3905,16 @@
int backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- ALOGV("%s: usecase = %d", __func__, usecase->id );
-
backend_idx = platform_get_backend_index(snd_device);
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);
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
+ ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
+ new_sample_rate, backend_idx, usecase->id,
+ platform_get_snd_device_name(snd_device));
+
if (platform_check_codec_backend_cfg(adev, usecase, snd_device,
&new_bit_width, &new_sample_rate)) {
platform_set_codec_backend_cfg(adev, snd_device,