Merge " hal: msm8974: Fix echo reference port for msm8x10"
diff --git a/hal/Android.mk b/hal/Android.mk
index 87ebeea..bc17a57 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -37,6 +37,10 @@
LOCAL_SRC_FILES += audio_extn/audio_extn.c
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_PCM_OFFLOAD)),true)
+ LOCAL_CFLAGS += -DPCM_OFFLOAD_ENABLED
+endif
+
ifneq ($(strip $(AUDIO_FEATURE_DISABLED_ANC_HEADSET)),true)
LOCAL_CFLAGS += -DANC_HEADSET_ENABLED
endif
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 7feeb42..80bc434 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -22,6 +22,45 @@
#include <cutils/str_parms.h>
+#ifndef PCM_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_PCM_OFFLOAD 0x17000000UL
+#define AUDIO_FORMAT_PCM_16_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_16_BIT)
+#define AUDIO_FORMAT_PCM_24_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_8_24_BIT)
+#define AUDIO_OFFLOAD_CODEC_FORMAT "music_offload_codec_format"
+static inline bool audio_is_offload_pcm(audio_format_t format) {
+ return ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD);
+}
+#endif
+
+#ifndef AFE_PROXY_ENABLED
+#define AUDIO_DEVICE_OUT_PROXY 0x40000
+#endif
+
+#ifndef COMPRESS_VOIP_ENABLED
+#define AUDIO_OUTPUT_FLAG_VOIP_RX 0x4000
+#endif
+
+#ifndef INCALL_MUSIC_ENABLED
+#define AUDIO_OUTPUT_FLAG_INCALL_MUSIC 0x8000
+#endif
+
+#ifndef FLUENCE_ENABLED
+#define AUDIO_PARAMETER_KEY_FLUENCE "fluence"
+#define AUDIO_PARAMETER_VALUE_QUADMIC "quadmic"
+#define AUDIO_PARAMETER_VALUE_DUALMIC "dualmic"
+#define AUDIO_PARAMETER_KEY_NO_FLUENCE "none"
+#endif
+
+#ifndef FM_ENABLED
+#define AUDIO_DEVICE_OUT_FM 0x80000
+#define AUDIO_DEVICE_OUT_FM_TX 0x100000
+#define AUDIO_SOURCE_FM_RX 9
+#define AUDIO_SOURCE_FM_RX_A2DP 10
+#define AUDIO_DEVICE_IN_FM_RX (AUDIO_DEVICE_BIT_IN | 0x8000)
+#define AUDIO_DEVICE_IN_FM_RX_A2DP AUDIO_DEVICE_BIT_IN | 0x10000
+#endif
+
+
void audio_extn_set_parameters(struct audio_device *adev,
struct str_parms *parms);
diff --git a/hal/audio_extn/fm.c b/hal/audio_extn/fm.c
index a4157f8..35b20b8 100644
--- a/hal/audio_extn/fm.c
+++ b/hal/audio_extn/fm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -64,7 +64,7 @@
{
int32_t vol, ret = 0;
struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "Internal FM RX Volume";
+ const char *mixer_ctl_name = FM_RX_VOLUME;
ALOGV("%s: entry", __func__);
ALOGD("%s: (%f)\n", __func__, value);
@@ -92,7 +92,6 @@
return -EINVAL;
}
mixer_ctl_set_value(ctl, 0, vol);
-
ALOGV("%s: exit", __func__);
return ret;
}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9f76d31..2f67784 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -300,10 +300,12 @@
* control in use-case does not work because rate update takes place after
* AFE port open due to the limitation of mixer control order execution.
*/
- if (snd_device == SND_DEVICE_OUT_BT_SCO) {
+ if ((snd_device == SND_DEVICE_OUT_BT_SCO) ||
+ (snd_device == SND_DEVICE_IN_BT_SCO_MIC)) {
audio_route_apply_path(adev->audio_route, BT_SCO_SAMPLE_RATE);
audio_route_update_mixer(adev->audio_route);
- } else if (snd_device == SND_DEVICE_OUT_BT_SCO_WB) {
+ } else if ((snd_device == SND_DEVICE_OUT_BT_SCO_WB) ||
+ (snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB)) {
audio_route_apply_path(adev->audio_route, BT_SCO_WB_SAMPLE_RATE);
audio_route_update_mixer(adev->audio_route);
}
@@ -437,23 +439,22 @@
/* Make sure all the streams are de-routed before disabling the device */
audio_route_update_mixer(adev->audio_route);
+ /* Make sure the previous devices to be disabled first and then enable the
+ selected devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- disable_snd_device(adev, usecase->out_snd_device, false);
+ disable_snd_device(adev, usecase->out_snd_device, true);
}
}
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- enable_snd_device(adev, snd_device, false);
+ enable_snd_device(adev, snd_device, true);
}
}
- /* Make sure new snd device is enabled before re-routing the streams */
- audio_route_update_mixer(adev->audio_route);
-
/* Re-route all the usecases on the shared backend other than the
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
@@ -507,23 +508,22 @@
/* Make sure all the streams are de-routed before disabling the device */
audio_route_update_mixer(adev->audio_route);
+ /* Make sure the previous devices to be disabled first and then enable the
+ selected devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- disable_snd_device(adev, usecase->in_snd_device, false);
+ disable_snd_device(adev, usecase->in_snd_device, true);
}
}
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
- enable_snd_device(adev, snd_device, false);
+ enable_snd_device(adev, snd_device, true);
}
}
- /* Make sure new snd device is enabled before re-routing the streams */
- audio_route_update_mixer(adev->audio_route);
-
/* Re-route all the usecases on the shared backend other than the
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
@@ -637,7 +637,8 @@
}
} else if (voice_extn_compress_voip_is_active(adev)) {
voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
- if (voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
+ if ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) {
in_snd_device = voip_usecase->in_snd_device;
out_snd_device = voip_usecase->out_snd_device;
}
@@ -698,12 +699,12 @@
/* Disable current sound devices */
if (usecase->out_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase, true);
- disable_snd_device(adev, usecase->out_snd_device, false);
+ disable_snd_device(adev, usecase->out_snd_device, true);
}
if (usecase->in_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase, true);
- disable_snd_device(adev, usecase->in_snd_device, false);
+ disable_snd_device(adev, usecase->in_snd_device, true);
}
/* Applicable only on the targets that has external modem.
@@ -1424,13 +1425,14 @@
pthread_mutex_lock(&adev->lock);
/*
- * When HDMI cable is unplugged the music playback is paused and
- * the policy manager sends routing=0. But the audioflinger
- * continues to write data until standby time (3sec).
- * As the HDMI core is turned off, the write gets blocked.
+ * When HDMI cable is unplugged/usb hs is disconnected the
+ * music playback is paused and the policy manager sends routing=0
+ * But the audioflingercontinues to write data until standby time
+ * (3sec). As the HDMI core is turned off, the write gets blocked.
* Avoid this by routing audio to speaker until standby.
*/
- if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL &&
+ if ((out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+ out->devices == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET) &&
val == AUDIO_DEVICE_NONE) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
@@ -2255,7 +2257,8 @@
/* Check if this usecase is already existing */
pthread_mutex_lock(&adev->lock);
- if (get_usecase_from_list(adev, out->usecase) != NULL) {
+ if ((get_usecase_from_list(adev, out->usecase) != NULL) &&
+ (out->usecase != USECASE_COMPRESS_VOIP_CALL)) {
ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
pthread_mutex_unlock(&adev->lock);
ret = -EEXIST;
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index 7b955ba..661a0d0 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -137,6 +137,12 @@
hw_info->snd_devices = NULL;
hw_info->num_snd_devices = 0;
strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8x16-skui-snd-card")) {
+ strlcpy(hw_info->type, "skui", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8x16", sizeof(hw_info->name));
+ hw_info->snd_devices = NULL;
+ hw_info->num_snd_devices = 0;
+ strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
} else {
ALOGW("%s: Not an 8x16 device", __func__);
}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index f081912..f12697c 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -178,6 +178,8 @@
#define PLAYBACK_OFFLOAD_DEVICE 9
#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
+/* Define macro for Internal FM volume mixer */
+#define FM_RX_VOLUME "Internal FM RX Volume"
#define LOWLATENCY_PCM_DEVICE 12
#define EC_REF_RX "I2S_RX"
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index e38d801..e326785 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -118,4 +118,7 @@
#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
#define AUDIO_CAPTURE_PERIOD_COUNT 2
+/* Define macro for Internal FM volume mixer */
+#define FM_RX_VOLUME "Internal FM RX Volume"
+
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a303a30..5315e78 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1433,6 +1433,7 @@
}
} else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
+ set_echo_reference(adev->mixer, EC_REF_RX);
} else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
@@ -1454,6 +1455,7 @@
}
} else {
snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
+ set_echo_reference(adev->mixer, EC_REF_RX);
}
}
} else if (source == AUDIO_SOURCE_CAMCORDER) {
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index d0b6418..f86faf6 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -221,7 +221,7 @@
#define VOICE_CALL_PCM_DEVICE 20
#define VOICE2_CALL_PCM_DEVICE 25
#define VOLTE_CALL_PCM_DEVICE 21
-#define QCHAT_CALL_PCM_DEVICE 06
+#define QCHAT_CALL_PCM_DEVICE 33
#define VOWLAN_CALL_PCM_DEVICE -1
#elif PLATFORM_MSM8610
#define VOICE_CALL_PCM_DEVICE 2
@@ -245,6 +245,12 @@
#define HFP_ASM_RX_TX 24
#endif
+#ifdef PLATFORM_APQ8084
+#define FM_RX_VOLUME "Quat MI2S FM RX Volume"
+#else
+#define FM_RX_VOLUME "Internal FM RX Volume"
+#endif
+
#define LIB_CSD_CLIENT "libcsd-client.so"
/* CSD-CLIENT related functions */
typedef int (*init_t)(bool);
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index f6083f3..5dbd7b9 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -570,6 +570,7 @@
voice_extn_compress_voip_in_get_parameters(in, query, reply);
}
+#ifdef INCALL_MUSIC_ENABLED
int voice_extn_check_and_set_incall_music_usecase(struct audio_device *adev,
struct stream_out *out)
{
@@ -591,4 +592,5 @@
return 0;
}
+#endif
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index 4f3a737..517f207 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -1,4 +1,4 @@
-ifeq ($(strip $(BOARD_USES_ALSA_AUDIO)),true)
+ifneq ($(strip $(BOARD_USES_AOSP_AUDIO_POLICY_MANAGER)),true)
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index a6f0b0b..03f6c41 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -595,7 +595,63 @@
return 0;
}
+#ifdef VOICE_CONCURRENCY
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.record.conc.disabled", propValue, NULL)) {
+ prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if (prop_rec_enabled) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState) &&
+ (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("Creating input during incall mode for inputSource: %d ",inputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if(prop_voip_enabled) {
+ ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+ return 0;
+ }
+ break;
+
+ default:
+ ALOGD("BLOCKING input during incall mode for inputSource: %d ",inputSource);
+ return 0;
+ }
+ }
+ }//check for VoIP flag
+ else if(prop_voip_enabled) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState) &&
+ (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+ ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+ return 0;
+ }
+ }
+ }
+
+#endif
IOProfile *profile = getInputProfile(device,
samplingRate,
format,
@@ -1076,19 +1132,11 @@
if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::BLUETOOTH_SCO) {
float voiceVolume;
-
- voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
-
- // Force voice volume to max when Vgs is set for bluetooth SCO as volume is managed by the headset
- if (stream == AudioSystem::BLUETOOTH_SCO) {
- String8 key ("bt_headset_vgs");
- mpClientInterface->getParameters(output,key);
- AudioParameter result(mpClientInterface->getParameters(0,key));
- int value;
- if (result.getInt(String8("isVGS"),value) == NO_ERROR) {
- ALOGV("Use BT-SCO Voice Volume");
- voiceVolume = 1.0;
- }
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AudioSystem::VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else {
+ voiceVolume = 1.0;
}
if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
@@ -1148,13 +1196,49 @@
IOProfile *profile = NULL;
#ifdef VOICE_CONCURRENCY
- if (isInCall()) {
- ALOGV(" IN call mode adding ULL flags .. flags: %x ", flags );
- //For voip paths
- if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
- flags = AudioSystem::OUTPUT_FLAG_DIRECT;
- else //route every thing else to ULL path
- flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_play_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
+ prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if (prop_play_enabled) {
+ //check if voice call is active / running in background
+ if((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState)
+ && (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ if(prop_voip_enabled) {
+ ALOGD(" IN call mode returing no output .. for VoIP usecase flags: %x ", flags );
+ // flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ return 0;
+ }
+ }
+ else {
+ ALOGD(" IN call mode adding ULL flags .. flags: %x ", flags );
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ }
+ }
+ } else if (prop_voip_enabled) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //return only ULL ouput
+ if((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState)
+ && (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ ALOGD(" IN call mode returing no output .. for VoIP usecase flags: %x ", flags );
+ // flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ return 0;
+ }
+ }
}
#endif
@@ -1337,11 +1421,18 @@
offloadInfo.has_video);
#ifdef VOICE_CONCURRENCY
- if(isInCall())
- {
- ALOGD("\n blocking compress offload on call mode\n");
- return false;
+ char concpropValue[PROPERTY_VALUE_MAX];
+ if(property_get("voice.playback.conc.disabled", concpropValue, NULL)) {
+ bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4);
+ if (propenabled) {
+ if(isInCall())
+ {
+ ALOGD("\n blocking compress offload on call mode\n");
+ return false;
+ }
+ }
}
+
#endif
// Check if stream type is music, then only allow offload as of now.
if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
@@ -1449,7 +1540,7 @@
void AudioPolicyManager::setPhoneState(int state)
{
- ALOGV("setPhoneState() state %d", state);
+ ALOGD("setPhoneState() state %d", state);
audio_devices_t newDevice = AUDIO_DEVICE_NONE;
if (state < 0 || state >= AudioSystem::NUM_MODES) {
ALOGW("setPhoneState() invalid state %d", state);
@@ -1514,6 +1605,117 @@
if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
newDevice = hwOutputDesc->device();
}
+#ifdef VOICE_CONCURRENCY
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
+ prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.record.conc.disabled", propValue, NULL)) {
+ prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if((AudioSystem::MODE_IN_CALL != oldState) && (AudioSystem::MODE_IN_CALL == state)) {
+ ALOGD("Entering to call mode oldState :: %d state::%d ",oldState, state);
+
+ if(prop_playback_enabled) {
+ //Call invalidate to reset all opened non ULL audio tracks
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ ALOGV(" Invalidate on call mode for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+ 0 /* ignored */);
+ }
+ }
+
+ if(prop_rec_enabled) {
+ //Close all active inputs
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+ switch(activeDesc->mInputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("FOUND active input during call active: %d",activeDesc->mInputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if(prop_voip_enabled) {
+ ALOGD("CLOSING VoIP input source on call setup :%d ",activeDesc->mInputSource);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ }
+ break;
+
+ default:
+ ALOGD("CLOSING input on call setup for inputSource: %d",activeDesc->mInputSource);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ break;
+ }
+ }
+ } else if(prop_voip_enabled) {
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+ if(AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) {
+ ALOGD("CLOSING VoIP on call setup : %d",activeDesc->mInputSource);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ }
+ }
+ }
+
+ //suspend PCM (deep-buffer) output & close compress & direct tracks
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY))
+ && prop_playback_enabled) {
+ ALOGD(" calling suspendOutput on call mdoe for primary output");
+ mpClientInterface->suspendOutput(mOutputs.keyAt(i));
+ } //Close compress all sessions
+ else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+ && prop_playback_enabled) {
+ ALOGD(" calling closeOutput on call mdoe for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)
+ && prop_voip_enabled) {
+ ALOGD(" calling closeOutput on call mdoe for DIRECT output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+
+ if((AudioSystem::MODE_IN_CALL == oldState) && (AudioSystem::MODE_IN_CALL != state)
+ && prop_playback_enabled) {
+ ALOGD("EXITING from call mode oldState :: %d state::%d \n",oldState, state);
+ //restore PCM (deep-buffer) output after call termination
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ ALOGD("calling restoreOutput after call mode for primary output");
+ mpClientInterface->restoreOutput(mOutputs.keyAt(i));
+ }
+ }
+ //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+ for (int i = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ ALOGD("Invalidate on call mode for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+ 0 /* ignored */);
+ }
+ }
+ mPrevPhoneState = oldState;
+#endif
int delayMs = 0;
if (isStateInCall(state)) {
@@ -1560,21 +1762,7 @@
} else {
mLimitRingtoneVolume = false;
}
-
-#ifdef VOICE_CONCURRENCY
- //Call invalidate to reset all opened non ULL audio tracks
- if(isInCall())
- {
- // Move tracks associated to this strategy from previous output to new output
- for (int i = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- ALOGV("\n Invalidate on call mode for stream :: %d \n", i);
- //FIXME see fixme on name change
- mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
- 0 /* ignored */);
- }
- }
-#endif
-
+ ALOGD(" End of setPhoneState ... mPhoneState: %d ",mPhoneState);
}
extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index ca7031d..f2488b2 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -106,8 +106,12 @@
//parameter indicates if HDMI plug in/out detected
bool mHdmiAudioEvent;
+
private:
void handleNotificationRoutingForStream(AudioSystem::stream_type stream);
+ // Used for voip + voice concurrency usecase
+ int mPrevPhoneState;
+
};
};
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 2cb910c..c50b7d0 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -3,6 +3,10 @@
include $(CLEAR_VARS)
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_PROXY_DEVICE)),true)
+ LOCAL_CFLAGS += -DAFE_PROXY_ENABLED
+endif
+
LOCAL_SRC_FILES:= \
bundle.c \
equalizer.c \
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index 657195d..341f145 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -100,12 +100,12 @@
switch (param) {
case BASSBOOST_PARAM_STRENGTH_SUPPORTED:
- ALOGV("%s: BASSBOOST_PARAM_STRENGTH_SUPPORTED", __func__);
+ ALOGV("%s: BASSBOOST_PARAM_STRENGTH_SUPPORTED", __func__);
*(uint32_t *)value = 1;
break;
case BASSBOOST_PARAM_STRENGTH:
- ALOGV("%s: BASSBOOST_PARAM_STRENGTH", __func__);
+ ALOGV("%s: BASSBOOST_PARAM_STRENGTH", __func__);
*(int16_t *)value = bassboost_get_strength(bass_ctxt);
break;
@@ -133,7 +133,7 @@
switch (param) {
case BASSBOOST_PARAM_STRENGTH:
- ALOGV("%s BASSBOOST_PARAM_STRENGTH", __func__);
+ ALOGV("%s BASSBOOST_PARAM_STRENGTH", __func__);
strength = (uint32_t)(*(int16_t *)value);
bassboost_set_strength(bass_ctxt, strength);
break;
@@ -154,7 +154,9 @@
if((device == AUDIO_DEVICE_OUT_SPEAKER) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER) ||
+#ifdef AFE_PROXY_ENABLED
(device == AUDIO_DEVICE_OUT_PROXY) ||
+#endif
(device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
(device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET)) {
if (offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass))) {
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 54bca07..205b250 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -100,12 +100,12 @@
switch (param) {
case VIRTUALIZER_PARAM_STRENGTH_SUPPORTED:
- ALOGV("%s: VIRTUALIZER_PARAM_STRENGTH_SUPPORTED", __func__);
+ ALOGV("%s: VIRTUALIZER_PARAM_STRENGTH_SUPPORTED", __func__);
*(uint32_t *)value = 1;
break;
case VIRTUALIZER_PARAM_STRENGTH:
- ALOGV("%s: VIRTUALIZER_PARAM_STRENGTH", __func__);
+ ALOGV("%s: VIRTUALIZER_PARAM_STRENGTH", __func__);
*(int16_t *)value = virtualizer_get_strength(virt_ctxt);
break;
@@ -133,7 +133,7 @@
switch (param) {
case VIRTUALIZER_PARAM_STRENGTH:
- ALOGV("%s VIRTUALIZER_PARAM_STRENGTH", __func__);
+ ALOGV("%s VIRTUALIZER_PARAM_STRENGTH", __func__);
strength = (uint32_t)(*(int16_t *)value);
virtualizer_set_strength(virt_ctxt, strength);
break;
@@ -154,7 +154,9 @@
if((device == AUDIO_DEVICE_OUT_SPEAKER) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER) ||
+#ifdef AFE_PROXY_ENABLED
(device == AUDIO_DEVICE_OUT_PROXY) ||
+#endif
(device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
(device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET)) {
if (offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt))) {