hal: native audio backward compatibility
- all changes for native audio backward
compatibility and related.
- add set parameters from UI
- enable dynamic device switching
- featurize the code
- various backend concurrency changes.
Change-Id: Id0f824c4b4c033f42008a4e8868652c8f6fe5c42
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index d8484e1..26c676a 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -174,7 +174,16 @@
typedef struct codec_backend_cfg {
uint32_t sample_rate;
uint32_t bit_width;
-}codec_backend_t;
+ 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;
@@ -208,7 +217,7 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
- codec_backend_t backend_cfg[MAX_PORT];
+ codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
};
static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -393,8 +402,8 @@
[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,
+ [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1] = 130,
+ [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2] = 130,
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
@@ -887,7 +896,7 @@
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.
@@ -1313,11 +1322,23 @@
audio_extn_dev_arbi_init();
/* initialize backend config */
- for (idx = 0; idx < MAX_PORT; idx++) {
- my_data->backend_cfg[idx].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- my_data->backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ 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;
}
@@ -1583,6 +1604,111 @@
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)
{
@@ -1994,9 +2120,10 @@
snd_device = SND_DEVICE_OUT_ANC_FB_HEADSET;
else
snd_device = SND_DEVICE_OUT_ANC_HEADSET;
- } else if (OUTPUT_SAMPLING_RATE_44100 == sample_rate)
+ } else if (platform_get_native_support() &&
+ OUTPUT_SAMPLING_RATE_44100 == sample_rate)
snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
- else
+ else
snd_device = SND_DEVICE_OUT_HEADPHONES;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
if (my_data->external_spk_1)
@@ -2550,11 +2677,11 @@
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;
int err, ret;
- struct stream_out out;
if(value == NULL || platform == NULL || parms == NULL) {
ALOGE("[%s] received null pointer, failed",__func__);
goto done_key_audcal;
@@ -2704,7 +2831,7 @@
/* 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)
@@ -2816,6 +2943,7 @@
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};
@@ -2824,7 +2952,6 @@
char *rparms=NULL;
int ret=0, err;
uint32_t param_len;
- struct stream_out out;
if(query==NULL || platform==NULL || reply==NULL) {
ALOGE("[%s] received null pointer",__func__);
@@ -2947,6 +3074,7 @@
/* 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);
@@ -3091,43 +3219,39 @@
}
int platform_set_codec_backend_cfg(struct audio_device* adev,
- struct audio_usecase *usecase,
+ snd_device_t snd_device,
unsigned int bit_width, unsigned int sample_rate)
{
int ret = 0;
- char backend_port = ALL_CODEC_BACKEND_PORT;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- ALOGV("%s bit width: %d, sample rate: %d", __func__, bit_width, sample_rate);
-
- if(usecase->stream.out->devices & SND_DEVICE_OUT_HEADPHONES_44_1)
- backend_port = HEADPHONE_44_1_BACKEND_PORT;
-
- // form the mixer string with appropriate backend port
- char * mixer_ctl_name = "SLIM_";
- strlcat(mixer_ctl_name, &backend_port, sizeof(mixer_ctl_name));
- strlcat(mixer_ctl_name, "_RX Format", sizeof(mixer_ctl_name));
- ALOGV("%s: Format mixer command - %s", __func__, mixer_ctl_name);
+ 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->backend_cfg[(int)backend_port].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;
}
- my_data->backend_cfg[(int)backend_port].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);
}
/*
@@ -3138,15 +3262,8 @@
* 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
- // form the mixer string with appropriate backend port
- mixer_ctl_name = "SLIM_";
- strlcat(mixer_ctl_name, &backend_port, sizeof(mixer_ctl_name));
- strlcat(mixer_ctl_name, "_RX SampleRate", sizeof(mixer_ctl_name));
- ALOGV("%s: SampleRate mixer command - %s", __func__, mixer_ctl_name);
-
if (sample_rate !=
- my_data->backend_cfg[(int)backend_port].sample_rate) {
+ my_data->current_backend_cfg[backend_idx].sample_rate) {
char *rate_str = NULL;
struct mixer_ctl *ctl;
@@ -3156,10 +3273,12 @@
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:
@@ -3174,37 +3293,47 @@
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);
- my_data->backend_cfg[(int)backend_port].sample_rate = 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};
- char backend_port = ALL_CODEC_BACKEND_PORT;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- if(usecase->stream.out->devices & SND_DEVICE_OUT_HEADPHONES_44_1)
- backend_port = HEADPHONE_44_1_BACKEND_PORT;
+ 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) {
@@ -3224,51 +3353,63 @@
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 != my_data->backend_cfg[(int)backend_port].bit_width) ||
- (sample_rate != my_data->backend_cfg[(int)backend_port].sample_rate)) {
+ 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;
@@ -3279,25 +3420,27 @@
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)
{
- unsigned int new_bit_width, old_bit_width;
- unsigned int new_sample_rate, old_sample_rate;
- char backend_port = ALL_CODEC_BACKEND_PORT;
+ 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;
- ALOGV("platform_check_and_set_codec_backend_cfg usecase = %d",usecase->id );
+ ALOGV("%s: usecase = %d", __func__, usecase->id );
- if(usecase->stream.out->devices & SND_DEVICE_OUT_HEADPHONES_44_1)
- backend_port = HEADPHONE_44_1_BACKEND_PORT;
+ backend_idx = platform_get_backend_index(snd_device);
- new_bit_width = old_bit_width = my_data->backend_cfg[(int)backend_port].bit_width;
- new_sample_rate = old_sample_rate = my_data->backend_cfg[(int)backend_port].sample_rate;
+ new_bit_width = usecase->stream.out->bit_width;
+ new_sample_rate = usecase->stream.out->sample_rate;
- ALOGW("Codec backend bitwidth %d, samplerate %d", old_bit_width, old_sample_rate);
- if (platform_check_codec_backend_cfg(adev, usecase,
+ 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, usecase, new_bit_width, new_sample_rate);
+ platform_set_codec_backend_cfg(adev, snd_device,
+ new_bit_width, new_sample_rate);
return true;
}