Merge "audio: hal: Add support to send codec specific data in gapless"
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index aaca0b2..baec1e0 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -236,11 +236,17 @@
static bool is_external_codec = false;
static const int pcm_device_table_of_ext_codec[AUDIO_USECASE_MAX][2] = {
- [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC, QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC}
+ [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC, QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC},
+ [USECASE_VOICEMMODE1_CALL] = {VOICEMMODE1_CALL_PCM_DEVICE_OF_EXT_CODEC,
+ VOICEMMODE1_CALL_PCM_DEVICE_OF_EXT_CODEC},
+ [USECASE_VOICEMMODE2_CALL] = {VOICEMMODE2_CALL_PCM_DEVICE_OF_EXT_CODEC,
+ VOICEMMODE2_CALL_PCM_DEVICE_OF_EXT_CODEC},
};
/* List of use cases that has different PCM device ID's for internal and external codecs */
-static const int misc_usecase[AUDIO_USECASE_MAX] = { USECASE_QCHAT_CALL };
+static const int misc_usecase[AUDIO_USECASE_MAX] = {USECASE_QCHAT_CALL,
+ USECASE_VOICEMMODE1_CALL,
+ USECASE_VOICEMMODE2_CALL};
int pcm_device_table[AUDIO_USECASE_MAX][2] = {
[USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {DEEP_BUFFER_PCM_DEVICE,
@@ -281,6 +287,10 @@
[USECASE_VOLTE_CALL] = {VOLTE_CALL_PCM_DEVICE, VOLTE_CALL_PCM_DEVICE},
[USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE, QCHAT_CALL_PCM_DEVICE},
[USECASE_VOWLAN_CALL] = {VOWLAN_CALL_PCM_DEVICE, VOWLAN_CALL_PCM_DEVICE},
+ [USECASE_VOICEMMODE1_CALL] = {VOICEMMODE1_CALL_PCM_DEVICE,
+ VOICEMMODE1_CALL_PCM_DEVICE},
+ [USECASE_VOICEMMODE2_CALL] = {VOICEMMODE2_CALL_PCM_DEVICE,
+ VOICEMMODE2_CALL_PCM_DEVICE},
[USECASE_COMPRESS_VOIP_CALL] = {COMPRESS_VOIP_CALL_PCM_DEVICE, COMPRESS_VOIP_CALL_PCM_DEVICE},
[USECASE_INCALL_REC_UPLINK] = {AUDIO_RECORD_PCM_DEVICE,
AUDIO_RECORD_PCM_DEVICE},
@@ -656,6 +666,8 @@
{TO_NAME_INDEX(USECASE_VOLTE_CALL)},
{TO_NAME_INDEX(USECASE_QCHAT_CALL)},
{TO_NAME_INDEX(USECASE_VOWLAN_CALL)},
+ {TO_NAME_INDEX(USECASE_VOICEMMODE1_CALL)},
+ {TO_NAME_INDEX(USECASE_VOICEMMODE2_CALL)},
{TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK)},
{TO_NAME_INDEX(USECASE_INCALL_REC_DOWNLINK)},
{TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
@@ -742,7 +754,7 @@
if (!strncmp(snd_card_name, "msm8939-tapan-snd-card",
sizeof("msm8939-tapan-snd-card")) ||
!strncmp(snd_card_name, "msm8939-tapan9302-snd-card",
- sizeof("msm8939-tapan9302-snd-card"))||
+ sizeof("msm8939-tapan9302-snd-card")) ||
!strncmp(snd_card_name, "msm8939-tomtom9330-snd-card",
sizeof("msm8939-tomtom9330-snd-card")) ||
!strncmp(snd_card_name, "msm8952-tomtom-snd-card",
@@ -873,7 +885,7 @@
msm_be_id_array_len =
sizeof(msm_device_to_be_id_external_codec) / sizeof(msm_device_to_be_id_internal_codec[0]);
} else if (!strncmp(snd_card_name, "msm8952-snd-card-mtp",
- sizeof("msm8952-snd-card-mtpmsm8952-snd-card-mtp"))) {
+ sizeof("msm8952-snd-card-mtp"))) {
strlcpy(mixer_xml_path, MIXER_XML_PATH_MTP,
sizeof(MIXER_XML_PATH_MTP));
msm_device_to_be_id = msm_device_to_be_id_internal_codec;
@@ -1625,6 +1637,26 @@
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_5_RX SampleRate");
+
+ if (platform_get_native_support()) {
+
+ if (!strncmp(snd_card_name, "msm8976-tasha-snd-card",
+ sizeof("msm8976-tasha-snd-card")) ||
+ !strncmp(snd_card_name, "msm8976-tashalite-snd-card",
+ sizeof("msm8976-tashalite-snd-card"))) {
+ /* do nothing */
+ }
+ else {
+ platform_set_native_support(false);
+ }
+ }
+
+ ALOGD("native: native audio: %s for sound card %s",
+ (platform_get_native_support() ? "enabled" : "disabled"),
+ snd_card_name);
+
+
+
my_data->edid_info = NULL;
return my_data;
}
@@ -1914,7 +1946,7 @@
{
na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled
= codec_support;
- ALOGD("%s: na_props.platform_na_prop_enabled: %d", __func__,
+ ALOGV("%s: na_props.platform_na_prop_enabled: %d", __func__,
na_props.platform_na_prop_enabled);
return 0;
}
@@ -3502,6 +3534,8 @@
* USECASE_VOLTE_CALL:
* USECASE_QCHAT_CALL:
* USECASE_VOWLAN_CALL:
+ * USECASE_VOICEMMODE1_CALL:
+ * USECASE_VOICEMMODE2_CALL:
* USECASE_COMPRESS_VOIP_CALL:
* USECASE_AUDIO_RECORD_FM_VIRTUAL:
* USECASE_INCALL_REC_UPLINK:
@@ -3564,6 +3598,8 @@
* USECASE_VOLTE_CALL:
* USECASE_QCHAT_CALL:
* USECASE_VOWLAN_CALL:
+ * USECASE_VOICEMMODE1_CALL:
+ * USECASE_VOICEMMODE2_CALL:
* USECASE_COMPRESS_VOIP_CALL:
* USECASE_AUDIO_RECORD_FM_VIRTUAL:
* USECASE_INCALL_REC_UPLINK:
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 511d774..a5b97ac 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -264,6 +264,12 @@
#define QCHAT_CALL_PCM_DEVICE_OF_EXT_CODEC 28
#define VOWLAN_CALL_PCM_DEVICE 16
+#define VOICEMMODE1_CALL_PCM_DEVICE 26
+#define VOICEMMODE2_CALL_PCM_DEVICE 27
+
+#define VOICEMMODE1_CALL_PCM_DEVICE_OF_EXT_CODEC 29
+#define VOICEMMODE2_CALL_PCM_DEVICE_OF_EXT_CODEC 30
+
#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
#define AFE_PROXY_RECORD_PCM_DEVICE 8
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index a1becd8..b0707d7 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -907,6 +907,53 @@
mLimitRingtoneVolume = false;
}
}
+
+void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+
+ if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+ ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+ return;
+ }
+ bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ // Use reverse loop to make sure any low latency usecases (generally tones)
+ // are not routed before non LL usecases (generally music).
+ // We can safely assume that LL output would always have lower index,
+ // and use this work-around to avoid routing of output with music stream
+ // from the context of short lived LL output.
+ // Note: in case output's share backend(HAL sharing is implicit) all outputs
+ // gets routing update while processing first output itself.
+ for (size_t i = mOutputs.size(); i > 0; i--) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
+ audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || outputDesc != mPrimaryOutput) {
+ setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(outputDesc, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0) {
+ setInputDevice(activeInput, getNewInputDevice(activeInput));
+ }
+
+}
+
status_t AudioPolicyManagerCustom::stopSource(sp<SwAudioOutputDescriptor> outputDesc,
audio_stream_type_t stream,
bool forceDeviceUpdate)
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 55bb05c..dfaad65 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -54,6 +54,8 @@
const char *device_address,
const char *device_name);
virtual void setPhoneState(audio_mode_t state);
+ virtual void setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index d363b77..b2f0952 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -300,20 +300,40 @@
return false;
}
-int configure_proxy_capture(struct mixer *mixer, int value) {
- const char *proxy_ctl_name = "AFE_PCM_RX Audio Mixer MultiMedia4";
+int set_control(const char* name, struct mixer *mixer, int value) {
struct mixer_ctl *ctl;
+ ctl = mixer_get_ctl_by_name(mixer, name);
+ if (ctl == NULL) {
+ ALOGW("%s: could not get %s ctl", __func__, name);
+ return -EINVAL;
+ }
+ if (mixer_ctl_set_value(ctl, 0, value) != 0) {
+ ALOGW("%s: error setting value %d on %s ", __func__, value, name);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int configure_proxy_capture(struct mixer *mixer, int value) {
+ int retval = 0;
+
if (value && acdb_send_audio_cal)
acdb_send_audio_cal(AFE_PROXY_ACDB_ID, ACDB_DEV_TYPE_OUT);
- ctl = mixer_get_ctl_by_name(mixer, proxy_ctl_name);
- if (ctl == NULL) {
- ALOGW("%s: could not get %s ctl", __func__, proxy_ctl_name);
- return -EINVAL;
- }
- if (mixer_ctl_set_value(ctl, 0, value) != 0)
- ALOGW("%s: error setting value %d on %s ", __func__, value, proxy_ctl_name);
+ retval = set_control("AFE_PCM_RX Audio Mixer MultiMedia4", mixer, value);
+
+ if (retval != 0)
+ return retval;
+
+ // Extending visualizer to capture for compress2 path as well.
+ // for extending it to multiple offload either this needs to be extended
+ // or need to find better solution to enable only active offload sessions
+
+ retval = set_control("AFE_PCM_RX Audio Mixer MultiMedia7", mixer, value);
+ if (retval != 0)
+ return retval;
return 0;
}