Merge "audio-hal: fix variables used uninitialized"
diff --git a/configs/msmsteppe/mixer_paths_wcd9375.xml b/configs/msmsteppe/mixer_paths_wcd9375.xml
index 85de3eb..680f445 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375.xml
@@ -371,8 +371,9 @@
<ctl name="COMP7 Switch" value="0" />
<ctl name="COMP8 Switch" value="0" />
- <!-- Headphone Default mode - AB -->
- <ctl name="RX HPH Mode" value="CLS_AB" />
+ <!-- Headphone class-H mode -->
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
<!-- IIR/voice anc -->
<ctl name="IIR0 Band1" id ="0" value="268435456" />
@@ -2130,9 +2131,13 @@
</path>
<path name="hph-highquality-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
</path>
<path name="hph-lowpower-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
</path>
<path name="true-native-mode">
diff --git a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
index df6890b..2a9499e 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
@@ -371,8 +371,9 @@
<ctl name="COMP7 Switch" value="0" />
<ctl name="COMP8 Switch" value="0" />
- <!-- Headphone Default mode - AB -->
- <ctl name="RX HPH Mode" value="CLS_AB" />
+ <!-- Headphone class-H mode -->
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
<!-- IIR/voice anc -->
<ctl name="IIR0 Band1" id ="0" value="268435456" />
@@ -2210,9 +2211,13 @@
</path>
<path name="hph-highquality-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
</path>
<path name="hph-lowpower-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
</path>
<path name="true-native-mode">
diff --git a/configs/sdm710/audio_platform_info_intcodec.xml b/configs/sdm710/audio_platform_info_intcodec.xml
index 603ec57..4ae3bb1 100644
--- a/configs/sdm710/audio_platform_info_intcodec.xml
+++ b/configs/sdm710/audio_platform_info_intcodec.xml
@@ -27,6 +27,8 @@
<audio_platform_info>
<bit_width_configs>
<device name="SND_DEVICE_OUT_SPEAKER" bit_width="24"/>
+ <device name="SND_DEVICE_IN_HANDSET_MIC" bit_width="24"/>
+ <device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" bit_width="24"/>
</bit_width_configs>
<interface_names>
<device name="AUDIO_DEVICE_IN_BUILTIN_MIC" interface="INT3_MI2S" codec_type="internal"/>
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 8e62a02..133939f 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -7276,6 +7276,7 @@
in->usecase = USECASE_AUDIO_RECORD_MMAP;
in->config = pcm_config_mmap_capture;
in->config.format = pcm_format_from_audio_format(config->format);
+ in->config.channels = channel_count;
in->stream.start = in_start;
in->stream.stop = in_stop;
in->stream.create_mmap_buffer = in_create_mmap_buffer;
@@ -7427,8 +7428,10 @@
// between the callback and close_stream
audio_extn_snd_mon_unregister_listener(stream);
- /* Disable echo reference while closing input stream */
- platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
+ // Disable echo reference if there are no active input and hfp call
+ // while closing input stream
+ if (!adev->active_input && !audio_extn_hfp_is_active(adev))
+ platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
if (in == NULL) {
ALOGE("%s: audio_stream_in ptr is NULL", __func__);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 2539ed1..30b9d10 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -2381,8 +2381,6 @@
ALOGD("ACDB initialization failed");
}
}
- /* init keep-alive for compress passthru */
- audio_extn_keep_alive_init(adev);
#ifdef DYNAMIC_LOG_ENABLED
log_utils_init();
#endif
@@ -6473,7 +6471,8 @@
*/
static bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
int backend_idx,
- struct audio_backend_cfg *backend_cfg)
+ struct audio_backend_cfg *backend_cfg,
+ snd_device_t snd_device)
{
bool backend_change = false;
unsigned int bit_width;
@@ -6494,14 +6493,18 @@
// 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)
- || (my_data->is_internal_codec)) {
+ 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;
channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+ } else if (my_data->is_internal_codec && !audio_is_usb_in_device(snd_device)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+ if (adev->active_input->bit_width == 24)
+ bit_width = platform_get_snd_device_bit_width(snd_device);
} else {
struct listnode *node;
struct audio_usecase *uc = NULL;
@@ -6593,7 +6596,7 @@
backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
- &backend_cfg)) {
+ &backend_cfg, snd_device)) {
ret = platform_set_codec_backend_cfg(adev, snd_device,
backend_cfg);
if(!ret)