hal: changes to support 24 bit record
-Changes to support 24 bit record if input format request is
AUDIO_FORMAT_PCM_8_24_BIT or AUDIO_FORMAT_PCM_24_BIT_PACKED
Change-Id: I68076524ccccbf9f0be3c88bb01180ae7e4fd8b1
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 4b20544..385d20b 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -243,6 +243,7 @@
bool edid_valid;
char ec_ref_mixer_path[64];
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
+ codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
char codec_version[CODEC_VERSION_MAX_LENGTH];
int hw_dep_fd;
};
@@ -1630,6 +1631,11 @@
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_5_RX SampleRate");
+ my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_0_TX Format");
+ my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_0_TX SampleRate");
+
ret = audio_extn_utils_get_codec_version(snd_card_name,
my_data->adev->snd_card,
my_data->codec_version);
@@ -4075,6 +4081,202 @@
return ret;
}
+/*
+ * configures afe with bit width and Sample Rate
+ */
+
+int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
+ snd_device_t snd_device,
+ unsigned int bit_width, unsigned int sample_rate,
+ audio_format_t format)
+{
+ int ret = 0;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+
+ ALOGI("%s:txbecf: 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_tx_backend_cfg[backend_idx].bit_width) {
+
+ struct mixer_ctl *ctl = NULL;
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
+ if (!ctl) {
+ ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
+ __func__,
+ my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
+ return -EINVAL;
+ }
+ if (bit_width == 24) {
+ if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ else
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ } else {
+ ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
+ }
+
+ if (ret < 0) {
+ ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
+ __func__,
+ my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
+ return -EINVAL;
+ }
+
+ my_data->current_tx_backend_cfg[backend_idx].bit_width = bit_width;
+ ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
+ my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+ }
+
+ /*
+ * Backend sample rate configuration follows:
+ * 16 bit record - 48khz for streams at any valid sample rate
+ * 24 bit record - 48khz for stream sample rate less than 48khz
+ * 24 bit record - 96khz for sample rate range of 48khz to 96khz
+ * 24 bit record - 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 != my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
+ /*
+ * sample rate update is needed only for hifi audio enabled platforms
+ */
+ char *rate_str = NULL;
+ struct mixer_ctl *ctl = NULL;
+
+ switch (sample_rate) {
+ case 8000:
+ case 11025:
+ case 16000:
+ case 22050:
+ case 32000:
+ case 44100:
+ case 48000:
+ rate_str = "KHZ_48";
+ break;
+ case 64000:
+ case 88200:
+ case 96000:
+ rate_str = "KHZ_96";
+ break;
+ case 176400:
+ case 192000:
+ rate_str = "KHZ_192";
+ break;
+ default:
+ rate_str = "KHZ_48";
+ break;
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
+
+ if (!ctl) {
+ ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
+ __func__,
+ my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
+ return -EINVAL;
+ }
+
+ ALOGD("%s:txbecf: afe: %s set to %s", __func__,
+ my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
+ rate_str);
+ ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
+ if (ret < 0) {
+ ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
+ __func__,
+ my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
+ return -EINVAL;
+ }
+
+ my_data->current_tx_backend_cfg[backend_idx].sample_rate = sample_rate;
+ }
+
+ return ret;
+}
+
+/*
+ * goes through all the current usecases and picks the highest
+ * bitwidth & samplerate
+ */
+bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
+ unsigned int* new_bit_width,
+ unsigned int* new_sample_rate)
+{
+ bool backend_change = false;
+ unsigned int bit_width;
+ unsigned int sample_rate;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+
+ bit_width = *new_bit_width;
+ sample_rate = *new_sample_rate;
+
+ ALOGI("%s:txbecf: afe: 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 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) {
+ ALOGW("%s:txbecf: afe:Use default bw and sr for voice/voip calls and "
+ "for unprocessed/camera source", __func__);
+ bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ }
+
+ ALOGI("%s:txbecf: 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_tx_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
+ *new_bit_width = bit_width;
+ *new_sample_rate = sample_rate;
+ backend_change = true;
+ ALOGI("%s:txbecf: 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;
+}
+
+bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
+ struct audio_usecase *usecase, snd_device_t snd_device)
+{
+ unsigned int new_bit_width;
+ unsigned int new_sample_rate;
+ audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
+ int backend_idx = DEFAULT_CODEC_BACKEND;
+ int ret = 0;
+ if(usecase->type == PCM_CAPTURE) {
+ new_sample_rate = usecase->stream.in->sample_rate;
+ new_bit_width = usecase->stream.in->bit_width;
+ format = usecase->stream.in->format;
+ } else {
+ new_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ new_sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ }
+
+ ALOGI("%s:txbecf: 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_capture_codec_backend_cfg(adev, &new_bit_width,
+ &new_sample_rate)) {
+ ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
+ new_bit_width, new_sample_rate, format);
+ if(!ret)
+ return true;
+ }
+
+ return false;
+}
+
int platform_set_snd_device_backend(snd_device_t device, const char *backend_tag,
const char * hw_interface)
{
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index caa40d1..7fe7271 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -179,7 +179,7 @@
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define OUTPUT_SAMPLING_RATE_44100 44100
-
+#define MAX_CODEC_TX_BACKENDS 1
enum {
DEFAULT_CODEC_BACKEND,
SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,