Merge "hal: Fix aanc sequence during voice call"
diff --git a/configs/msm8996/mixer_paths_tasha.xml b/configs/msm8996/mixer_paths_tasha.xml
index 9f63413..64d6bfb 100644
--- a/configs/msm8996/mixer_paths_tasha.xml
+++ b/configs/msm8996/mixer_paths_tasha.xml
@@ -2278,6 +2278,9 @@
<path name="capture-fm">
</path>
+ <path name="aanc-path">
+ </path>
+
<path name="aanc-handset-mic">
<ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index efd275d..23c0f06 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -2293,6 +2293,12 @@
<path name="capture-fm">
</path>
+ <path name="aanc-path">
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
+ </path>
+
<path name="aanc-handset-mic">
<ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
@@ -2309,9 +2315,6 @@
<ctl name="ADC MUX7" value="DMIC" />
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="IIR0 INP0 MUX" value="DEC6" />
- <ctl name="ADC MUX10" value="DMIC" />
- <ctl name="DMIC MUX10" value="DMIC4" />
- <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
</path>
<!-- Dual MIC devices -->
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index 29212f9..555555d 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -2068,6 +2068,12 @@
<path name="capture-fm">
</path>
+ <path name="aanc-path">
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
+ </path>
+
<path name="aanc-handset-mic">
<ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
@@ -2084,9 +2090,6 @@
<ctl name="ADC MUX7" value="DMIC" />
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="IIR0 INP0 MUX" value="DEC6" />
- <ctl name="ADC MUX10" value="DMIC" />
- <ctl name="DMIC MUX10" value="DMIC4" />
- <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
</path>
<!-- Dual MIC devices -->
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index e82cd51..270f36e 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -1424,6 +1424,10 @@
if (voice_is_call_state_active(adev) ||
voice_extn_compress_voip_is_started(adev))
voice_set_sidetone(adev, usecase->out_snd_device, false);
+
+ /* Disable aanc only if voice call exists */
+ if (voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, usecase->out_snd_device, false);
}
/* Disable current sound devices */
@@ -1494,6 +1498,10 @@
enable_audio_route(adev, usecase);
if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
+ /* Enable aanc only if voice call exists */
+ if (voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, out_snd_device, true);
+
/* Enable sidetone only if other voice/voip call already exists */
if (voice_is_call_state_active(adev) ||
voice_extn_compress_voip_is_started(adev))
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 0a14629..a42f984 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -5623,6 +5623,14 @@
return 0;
}
+void platform_update_aanc_path(struct audio_device *adev __unused,
+ snd_device_t out_snd_device __unused,
+ bool enable __unused,
+ char *str __unused)
+{
+ return;
+}
+
bool platform_check_codec_dsd_support(void *platform __unused)
{
return false;
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index b5a4f11..e025772 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1315,6 +1315,14 @@
return 0;
}
+void platform_update_aanc_path(struct audio_device *adev __unused,
+ snd_device_t out_snd_device __unused,
+ bool enable __unused,
+ char *str __unused)
+{
+ return;
+}
+
bool platform_check_codec_dsd_support(void *platform __unused)
{
return false;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index f506130..d209359 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -5647,6 +5647,22 @@
return 0;
}
+void platform_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable,
+ char *str)
+{
+ ALOGD("%s: aanc out device(%d) mixer cmd = %s, enable = %d\n",
+ __func__, out_snd_device, str, enable);
+
+ if (enable)
+ audio_route_apply_and_update_path(adev->audio_route, str);
+ else
+ audio_route_reset_and_update_path(adev->audio_route, str);
+
+ return;
+}
+
static void make_cal_cfg(acdb_audio_cal_cfg_t* cal, int acdb_dev_id,
int acdb_device_type, int app_type, int topology_id,
int sample_rate, uint32_t module_id, uint32_t param_id, bool persist)
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 76d99b3..61f42de 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -157,6 +157,10 @@
snd_device_t out_snd_device,
bool enable,
char * str);
+void platform_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable,
+ char * str);
bool platform_supports_true_32bit();
bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device, snd_device_t cuurent_snd_device);
bool platform_check_codec_dsd_support(void *platform);
diff --git a/hal/voice.c b/hal/voice.c
index b84c7b7..ca3098b 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -95,6 +95,39 @@
return;
}
+static bool voice_is_aanc_device(snd_device_t out_device,
+ char *mixer_path)
+{
+ bool is_aanc_dev;
+
+ switch (out_device) {
+ case SND_DEVICE_OUT_ANC_HANDSET:
+ is_aanc_dev = true;
+ strlcpy(mixer_path, "aanc-path", MIXER_PATH_MAX_LENGTH);
+ break;
+ default:
+ is_aanc_dev = false;
+ break;
+ }
+
+ return is_aanc_dev;
+}
+
+void voice_check_and_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable)
+{
+ char mixer_path[MIXER_PATH_MAX_LENGTH];
+
+ ALOGV("%s: %s, out_snd_device: %d\n",
+ __func__, (enable ? "enable" : "disable"), out_snd_device);
+
+ if (voice_is_aanc_device(out_snd_device, mixer_path))
+ platform_update_aanc_path(adev, out_snd_device, enable, mixer_path);
+
+ return;
+}
+
int voice_stop_usecase(struct audio_device *adev, audio_usecase_t usecase_id)
{
int ret = 0;
@@ -125,6 +158,10 @@
if (!voice_is_call_state_active(adev))
voice_set_sidetone(adev, uc_info->out_snd_device, false);
+ /* Disable aanc only when no calls are active */
+ if (!voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, false);
+
ret = platform_stop_voice_call(adev->platform, session->vsid);
/* 1. Close the PCM devices */
@@ -229,6 +266,10 @@
pcm_start(session->pcm_tx);
pcm_start(session->pcm_rx);
+ /* Enable aanc only when no calls are active */
+ if (!voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, true);
+
/* Enable sidetone only when no calls are already active */
if (!voice_is_call_state_active(adev))
voice_set_sidetone(adev, uc_info->out_snd_device, true);
diff --git a/hal/voice.h b/hal/voice.h
index efe48d8..3ae42a8 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -97,5 +97,8 @@
void voice_set_sidetone(struct audio_device *adev,
snd_device_t out_snd_device,
bool enable);
+void voice_check_and_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable);
bool voice_is_call_state_active(struct audio_device *adev);
#endif //VOICE_H