hal: Add support for Native Audio(44.1kHz) on msm8952
- Changes for device/Usecase selection
- Add new and combo devices for headphones
- Combo device handling
- Add new backend and logic to handle multiple
backends
- Remove hardcoded mixers and select them
dynamically based on the device.
- Port all the above changes from msm8974
platfrom to msm8916
Change-Id: Ib9c9e8fcd6530e35700fa2369f2029cf2de3199f
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index a752f80..b54209c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -620,7 +620,7 @@
ALOGV("%s: backend_idx: %d,"
"usecase_backend_idx: %d, curr device: %s, usecase device:"
"%s", __func__, backend_idx, usecase_backend_idx, platform_get_snd_device_name(snd_device),
- platform_get_snd_device_name(usecase->out_snd_device));
+ platform_get_snd_device_name(usecase->out_snd_device));
if (usecase->type != PCM_CAPTURE &&
(usecase->out_snd_device != snd_device || force_routing) &&
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index b5b7cb7..48472b6 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -177,6 +177,20 @@
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;
@@ -211,6 +225,7 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
+ codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
};
static bool is_external_codec = false;
@@ -294,6 +309,7 @@
[SND_DEVICE_OUT_SPEAKER_WSA] = "wsa-speaker",
[SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
[SND_DEVICE_OUT_HEADPHONES] = "headphones",
+ [SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1] = "speaker-and-headphones-ext-1",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = "speaker-and-headphones-ext-2",
@@ -398,6 +414,7 @@
[SND_DEVICE_OUT_SPEAKER_WSA] = 135,
[SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
[SND_DEVICE_OUT_HEADPHONES] = 10,
+ [SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = 10,
@@ -504,6 +521,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2)},
@@ -1036,7 +1054,7 @@
backend_table[dev] = NULL;
}
for (dev = 0; dev < SND_DEVICE_MAX; dev++) {
- backend_bit_width_table[dev] = 16;
+ backend_bit_width_table[dev] = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
}
// TBD - do these go to the platform-info.xml file.
@@ -1058,6 +1076,7 @@
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)
@@ -1190,8 +1209,6 @@
const char *snd_card_name;
char mixer_xml_path[100],ffspEnable[PROPERTY_VALUE_MAX];
char *cvd_version = NULL;
-
- my_data = calloc(1, sizeof(struct platform_data));
if (!my_data) {
ALOGE("failed to allocate platform data");
return NULL;
@@ -1426,6 +1443,26 @@
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;
}
@@ -1682,14 +1719,107 @@
return backend_bit_width_table[snd_device];
}
-int platform_set_native_support(bool codec_support __unused)
+int platform_set_native_support(bool codec_support)
{
+ na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
+ = codec_support;
+ ALOGV("%s: na_props.platform_na_prop_enabled: %d", __func__,
+ na_props.platform_na_prop_enabled);
return 0;
}
-int platform_get_backend_index(snd_device_t snd_device __unused)
+int platform_get_native_support()
{
- return 0;
+ int ret;
+ if (na_props.platform_na_prop_enabled) {
+ ret = na_props.ui_na_prop_enabled;
+ } else {
+ ret = na_props.platform_na_prop_enabled;
+ }
+ ALOGV("%s: na_props.ui_na_prop_enabled: %d", __func__,
+ na_props.ui_na_prop_enabled);
+ return ret;
+}
+
+void native_audio_get_params(struct str_parms *query,
+ struct str_parms *reply,
+ char *value, int len)
+{
+ int ret;
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ value, len);
+ if (ret >= 0) {
+ if (na_props.platform_na_prop_enabled) {
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ na_props.ui_na_prop_enabled ? "true" : "false");
+ ALOGV("%s: na_props.ui_na_prop_enabled: %d", __func__,
+ na_props.ui_na_prop_enabled);
+ } else {
+ str_parms_add_str(reply, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ "false");
+ ALOGV("%s: native audio not supported: %d", __func__,
+ na_props.platform_na_prop_enabled);
+ }
+ }
+}
+
+int native_audio_set_params(struct platform_data *platform,
+ struct str_parms *parms, char *value, int len)
+{
+ int ret = 0;
+ struct audio_usecase *usecase;
+ struct listnode *node;
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO,
+ value, len);
+ if (ret >= 0) {
+ if (na_props.platform_na_prop_enabled) {
+ if (!strncmp("true", value, sizeof("true")))
+ na_props.ui_na_prop_enabled = true;
+ else
+ na_props.ui_na_prop_enabled = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_NATIVE_AUDIO);
+
+ /* Iterate through the usecase list and trigger device switch for
+ all the appropriate usecases */
+ list_for_each(node, &(platform->adev)->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+
+ if (is_offload_usecase(usecase->id) &&
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) &&
+ OUTPUT_SAMPLING_RATE_44100 == usecase->stream.out->sample_rate) {
+ select_devices(platform->adev, usecase->id);
+ ALOGV("%s: triggering dynamic device switch for usecase: "
+ "%d, device: %d", __func__, usecase->id,
+ usecase->stream.out->devices);
+ }
+ }
+ } else {
+ ALOGV("%s: native audio not supported: %d", __func__,
+ na_props.platform_na_prop_enabled);
+ }
+ }
+ return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
+{
+ int32_t port = DEFAULT_CODEC_BACKEND;
+
+ if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ if (backend_table[snd_device] != NULL &&
+ !strcmp(backend_table[snd_device], "headphones-44.1"))
+ port = HEADPHONE_44_1_BACKEND;
+ else
+ port = DEFAULT_CODEC_BACKEND;
+ } else {
+ ALOGV("%s: Invalid device - %d ", __func__, snd_device);
+ }
+
+ ALOGV("%s: backend port - %d", __func__, port);
+ return port;
}
int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
@@ -1985,6 +2115,7 @@
audio_mode_t mode = adev->mode;
snd_device_t snd_device = SND_DEVICE_NONE;
audio_devices_t devices = out->devices;
+ unsigned int sample_rate = out->sample_rate;
#ifdef RECORD_PLAY_CONCURRENCY
bool use_voip_out_devices = false;
bool prop_rec_play_enabled = false;
@@ -2105,7 +2236,10 @@
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) {
@@ -2716,6 +2850,7 @@
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;
@@ -2752,7 +2887,8 @@
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);
@@ -2845,6 +2981,7 @@
}
}
#endif
+ native_audio_set_params(platform, parms, value, sizeof(value));
ALOGV("%s: exit with code(%d)", __func__, ret);
return ret;
}
@@ -2951,6 +3088,7 @@
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};
@@ -2981,7 +3119,8 @@
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) {
@@ -3077,6 +3216,7 @@
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VOLUME_BOOST, value);
}
+ native_audio_get_params(query, reply, value, sizeof(value));
/* Handle audio calibration keys */
kv_pairs = str_parms_to_str(reply);
@@ -3306,26 +3446,59 @@
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")))
+ {
+ /* external codec, for rest/old of the external codecs
+ we dont support this funtionality(chaning AFE params)
+ at the monment
+ */
+ return 1;
+ }
+ else {
+ /* internal codec */
+ return 0;
+ }
+}
+
int platform_set_codec_backend_cfg(struct audio_device* adev,
+ snd_device_t snd_device,
unsigned int bit_width, unsigned int sample_rate)
{
- ALOGV("%s bit width: %d, sample rate: %d", __func__, bit_width, sample_rate);
-
int ret = 0;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+ ALOGV("%s bit width: %d, sample rate: %d\n", __func__, bit_width, sample_rate);
+
const char *snd_card_name = mixer_get_name(adev->mixer);
- if (bit_width != adev->cur_codec_backend_bit_width) {
- const char * mixer_ctl_name;
- if (!strncmp(snd_card_name, "msm8952-tomtom-snd-card",
- sizeof("msm8952-tomtom-snd-card"))) {
- mixer_ctl_name = "SLIM_0_RX Format";
- }
- else
- mixer_ctl_name = "MI2S_RX Format";
+ int is_external_codec = platform_is_external_codec(snd_card_name);
+
+
+
+ backend_idx = platform_get_backend_index(snd_device);
+ ALOGV("%s bit width: %d, sample rate: %d backend_idx - %d\n",
+ __func__, bit_width, sample_rate, backend_idx);
+
+ if (bit_width !=
+ my_data->current_backend_cfg[backend_idx].bit_width) {
+
struct mixer_ctl *ctl;
- ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!is_external_codec) {
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ "MI2S_RX Format");
+
+ } else {
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
+ }
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer command - %s",
- __func__, mixer_ctl_name);
+ __func__, my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
return -EINVAL;
}
@@ -3333,10 +3506,12 @@
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);
}
/*
@@ -3348,9 +3523,12 @@
* 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) {
@@ -3359,10 +3537,12 @@
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:
@@ -3377,31 +3557,46 @@
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
@@ -3422,36 +3617,50 @@
list_for_each(node, &adev->usecase_list) {
struct audio_usecase *curr_usecase;
curr_usecase = node_to_item(node, struct audio_usecase, list);
- if (curr_usecase->type == PCM_PLAYBACK) {
+ if (curr_usecase->type == PCM_PLAYBACK &&
+ usecase != curr_usecase) {
struct stream_out *out =
(struct stream_out*) curr_usecase->stream.out;
- if (out != NULL ) {
- ALOGV("Offload playback running bw %d sr %d",
- out->bit_width, out->sample_rate);
+ usecase_backend_idx = platform_get_backend_index(curr_usecase->out_snd_device);
+
+ if (out != NULL &&
+ usecase_backend_idx == backend_idx) {
+ ALOGV("%s: usecase Offload playback running bw %d sr %d device %s be_idx %d",
+ __func__, out->bit_width, out->sample_rate,
+ platform_get_snd_device_name(curr_usecase->out_snd_device), usecase_backend_idx);
if (bit_width < out->bit_width)
bit_width = out->bit_width;
if (sample_rate < out->sample_rate)
sample_rate = out->sample_rate;
+ if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
}
}
}
+ if (backend_idx != HEADPHONE_44_1_BACKEND) {
+ // 16 bit playbacks are allowed through 16 bit/48 khz backend only for
+ // all non-native streams
+ if (16 == bit_width) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s: resetting sample_rate back to default, "
+ "backend_idx: %d", __func__, backend_idx);
+ }
- // 16 bit playback on speakers is allowed through 48 khz backend only
- if (16 == bit_width) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ // 24 bit playback on speakers is allowed through 48 khz backend only
+ // bit width re-configured based on platform info
+ if ((24 == bit_width) &&
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
+ bit_width = (uint32_t)platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ }
}
- // 24 bit playback on speakers is allowed through 48 khz backend only
- // bit width re-configured based on platform info
- if ((24 == bit_width) &&
- (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
- bit_width = (uint32_t)platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- }
+ ALOGI("%s Codec selected backend: %d updated bit width: %d and sample rate: %d",
+ __func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
- if ((bit_width != adev->cur_codec_backend_bit_width) ||
- (sample_rate != adev->cur_codec_backend_samplerate)) {
+ if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
*new_bit_width = bit_width;
*new_sample_rate = sample_rate;
backend_change = true;
@@ -3463,21 +3672,26 @@
}
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
- struct audio_usecase *usecase,
- snd_device_t snd_device __unused)
+ struct audio_usecase *usecase, snd_device_t snd_device)
{
- ALOGV("platform_check_and_set_codec_backend_cfg usecase = %d",usecase->id );
+ unsigned int new_bit_width;
+ unsigned int new_sample_rate;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
- unsigned int new_bit_width, old_bit_width;
- unsigned int new_sample_rate, old_sample_rate;
+ ALOGV("%s: usecase = %d", __func__, usecase->id );
- new_bit_width = old_bit_width = adev->cur_codec_backend_bit_width;
- new_sample_rate = old_sample_rate = adev->cur_codec_backend_samplerate;
+ backend_idx = platform_get_backend_index(snd_device);
- ALOGW("Codec backend bitwidth %d, samplerate %d", old_bit_width, old_sample_rate);
- if (platform_check_codec_backend_cfg(adev, usecase,
+ new_bit_width = usecase->stream.out->bit_width;
+ new_sample_rate = usecase->stream.out->sample_rate;
+
+ ALOGI("%s: Usecase bitwidth %d, samplerate %d, backend_idx %d",
+ __func__, new_bit_width, new_sample_rate, backend_idx);
+ if (platform_check_codec_backend_cfg(adev, usecase, snd_device,
&new_bit_width, &new_sample_rate)) {
- platform_set_codec_backend_cfg(adev, new_bit_width, new_sample_rate);
+ platform_set_codec_backend_cfg(adev, snd_device,
+ new_bit_width, new_sample_rate);
return true;
}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index aae05d0..e56bad1 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -37,7 +37,11 @@
/*
* Below are the devices for which is back end is same, SLIMBUS_0_RX.
* All these devices are handled by the internal HW codec. We can
- * enable any one of these devices at any time
+ * enable any one of these devices at any time. An exception here is
+ * 44.1k headphone which uses different backend. This is filtered
+ * as different hal internal device in the code but remains same
+ * as standard android device AUDIO_DEVICE_OUT_WIRED_HEADPHONE
+ * for other layers.
*/
#define AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND \
(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
@@ -163,13 +167,16 @@
};
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
-#define OUTPUT_SAMPLING_RATE_44100 44100
-
+#define OUTPUT_SAMPLING_RATE_44100 44100
+#define MAX_PORT 6
+#define ALL_CODEC_BACKEND_PORT 0
+#define HEADPHONE_44_1_BACKEND_PORT 5
enum {
DEFAULT_CODEC_BACKEND,
HEADPHONE_44_1_BACKEND,
MAX_CODEC_BACKENDS
};
+#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO "audio.nat.codec.enabled"
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION_MS 20