Merge 71514a7a7a85269cbd1686de896c5dedb6b82620 on remote branch
Change-Id: Ib3b35107efc55e48b9c43b5e5532dc253cda3ee5
diff --git a/configs/bengal/mixer_paths.xml b/configs/bengal/mixer_paths.xml
index 7829d5a..dbe85b7 100644
--- a/configs/bengal/mixer_paths.xml
+++ b/configs/bengal/mixer_paths.xml
@@ -1954,7 +1954,7 @@
<ctl name="MultiMedia17 Mixer SLIM_7_TX" value="1" />
</path>
- <path name="audio-record-compress bt-sco-wb">
+ <path name="audio-record-compress2 bt-sco-wb">
<ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-record-compress2 bt-sco" />
</path>
diff --git a/configs/msmnile_au/mixer_paths_sa8295_adp.xml b/configs/msmnile_au/mixer_paths_sa8295_adp.xml
index 7a3be75..3cc2250 100644
--- a/configs/msmnile_au/mixer_paths_sa8295_adp.xml
+++ b/configs/msmnile_au/mixer_paths_sa8295_adp.xml
@@ -279,9 +279,9 @@
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
<!-- hfp-sco -->
- <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia21" value="0" />
- <ctl name="MultiMedia21 Mixer AUX_PCM_UL_TX" value="0" />
- <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="0" />
+ <ctl name="PRI_TDM_RX_2 Audio Mixer MultiMedia21" value="0" />
+ <ctl name="MultiMedia21 Mixer SEN_TDM_TX_1" value="0" />
+ <ctl name="SEN_TDM_RX_1 Audio Mixer MultiMedia6" value="0" />
<ctl name="MultiMedia6 Mixer TERT_TDM_TX_0" value="0" />
<!-- icc-call and anc-loopback -->
@@ -297,7 +297,7 @@
<!-- EC Reference end -->
<path name="echo-reference">
- <ctl name="AUDIO_REF_EC_UL1 MUX" value="TERT_TDM_RX_2" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="PRI_TDM_RX_2" />
<ctl name="EC Reference Channels" value="One" />
<ctl name="EC Reference Bit Format" value="S16_LE" />
<ctl name="EC Reference SampleRate" value="48000" />
@@ -1708,6 +1708,11 @@
<path name="hfp-sco">
<path name="auto-record" />
+ <ctl name="SEN_TDM SlotWidth" value="16" />
+ <ctl name="SEN_TDM SlotNumber" value="Eight" />
+ <ctl name="SEN_TDM_RX_1 Channels" value="One" />
+ <ctl name="SEN_TDM_RX_1 SlotMapping" id ="0" value="0" />
+ <ctl name="SEN_TDM_RX_1 SlotMapping" id ="1" value="65535" />
<ctl name="SEN_TDM_RX_1 Audio Mixer MultiMedia6" value="1" />
<ctl name="MultiMedia6 Mixer TERT_TDM_TX_0" value="1" />
</path>
@@ -1723,7 +1728,7 @@
</path>
<path name="hfp-sco-wb">
- <ctl name="PRIM_AUX_PCM_RX SampleRate" value="KHZ_16" />
+ <ctl name="SEN_TDM_RX_1 SampleRate" value="KHZ_16" />
<path name="hfp-sco" />
</path>
@@ -1744,6 +1749,12 @@
<path name="hfp-sco-downlink">
<path name="auto-playback" />
+ <ctl name="SEN_TDM SlotWidth" value="16" />
+ <ctl name="SEN_TDM SlotNumber" value="Eight" />
+ <ctl name="SEN_TDM_TX_1 Channels" value="One" />
+ <ctl name="SEN_TDM_TX_1 SlotMapping" id ="0" value="0" />
+ <ctl name="SEN_TDM_TX_1 SlotMapping" id ="1" value="65535" />
+ <ctl name="PRI_TDM_RX_2 Channels" value="One" />
<ctl name="PRI_TDM_RX_2 SampleRate" value="KHZ_48" />
<ctl name="PRI_TDM_RX_2 SlotMapping" id ="0" value="28" />
<ctl name="PRI_TDM_RX_2 SlotMapping" id ="1" value="65535" />
@@ -1752,7 +1763,7 @@
</path>
<path name="hfp-sco-wb-downlink">
- <ctl name="PRIM_AUX_PCM_TX SampleRate" value="KHZ_16" />
+ <ctl name="SEN_TDM_TX_1 SampleRate" value="KHZ_16" />
<path name="hfp-sco-downlink" />
</path>
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 911a8d4..ff0558e 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -3127,6 +3127,38 @@
enable_audio_route(adev, voip_in_usecase);
}
}
+ if (voice_extn_compress_voip_is_active(adev)) {
+ struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
+ USECASE_COMPRESS_VOIP_CALL);
+ /*
+ * If only compress voip input is opened voip out will be primary out.
+ * Need to consider re-routing to select correct i/p pair
+ */
+ if ((voip_usecase != NULL) &&
+ (usecase->type == PCM_PLAYBACK) &&
+ (usecase->stream.out == voip_usecase->stream.out)) {
+ in_snd_device = platform_get_input_snd_device(adev->platform,
+ NULL,
+ &usecase->stream.out->device_list,
+ usecase->type);
+ if (voip_usecase->in_snd_device != in_snd_device ) {
+ ALOGD("%s:Re routing compress voip tx snd device matching voip rx pair",
+ __func__);
+ disable_audio_route(adev, voip_usecase);
+ disable_snd_device(adev, voip_usecase->in_snd_device);
+ voip_usecase->in_snd_device = in_snd_device;
+ voip_usecase->out_snd_device = usecase->out_snd_device;
+ /* Route all TX usecase to Compress voip BE */
+ check_usecases_capture_codec_backend(adev, voip_usecase, in_snd_device);
+ enable_snd_device(adev, in_snd_device);
+ /* Send Voice related calibration for RX /TX pair */
+ status = platform_switch_voice_call_device_post(adev->platform,
+ out_snd_device,
+ in_snd_device);
+ enable_audio_route(adev, voip_usecase);
+ }
+ }
+ }
audio_extn_qdsp_set_device(usecase);