diff --git a/packages/libpulseaudio/aaudio.patch b/packages/libpulseaudio/aaudio.patch
new file mode 100644
index 000000000..2ff743f29
--- /dev/null
+++ b/packages/libpulseaudio/aaudio.patch
@@ -0,0 +1,17 @@
+diff --git a/src/Makefile.am~ b/src/Makefile.am
+index f4464d2..eebcf02 100644
+--- a/src/Makefile.am~
++++ b/src/Makefile.am
+@@ -1495,6 +1495,12 @@ bin_SCRIPTS += utils/qpaeq
+ endif
+ endif
+ 
++modlibexec_LTLIBRARIES += module-aaudio-sink.la
++module_aaudio_sink_la_SOURCES = modules/aaudio/module-aaudio-sink.c
++module_aaudio_sink_la_LDFLAGS = $(MODULE_LDFLAGS) -laaudio
++module_aaudio_sink_la_LIBADD = $(MODULE_LIBADD)
++module_aaudio_sink_la_CFLAGS = $(AM_CFLAGS) -DPA_MODULE_NAME=module_aaudio_sink
++
+ # Simple protocol
+ 
+ module_simple_protocol_tcp_la_SOURCES = modules/module-protocol-stub.c
diff --git a/packages/libpulseaudio/build.sh b/packages/libpulseaudio/build.sh
index cc614bca1..1bbb0f81e 100644
--- a/packages/libpulseaudio/build.sh
+++ b/packages/libpulseaudio/build.sh
@@ -1,7 +1,7 @@
 TERMUX_PKG_HOMEPAGE=https://www.freedesktop.org/wiki/Software/PulseAudio
 TERMUX_PKG_DESCRIPTION="A featureful, general-purpose sound server - shared libraries"
 TERMUX_PKG_VERSION=12.2
-TERMUX_PKG_REVISION=4
+TERMUX_PKG_REVISION=5
 TERMUX_PKG_SHA256=809668ffc296043779c984f53461c2b3987a45b7a25eb2f0a1d11d9f23ba4055
 TERMUX_PKG_SRCURL=https://www.freedesktop.org/software/pulseaudio/releases/pulseaudio-${TERMUX_PKG_VERSION}.tar.xz
 TERMUX_PKG_DEPENDS="libltdl, libsndfile, libandroid-glob, libsoxr"
@@ -23,6 +23,8 @@ TERMUX_PKG_CONFFILES="etc/pulse/client.conf etc/pulse/daemon.conf etc/pulse/defa
 termux_step_pre_configure () {
 	mkdir $TERMUX_PKG_SRCDIR/src/modules/sles
 	cp $TERMUX_PKG_BUILDER_DIR/module-sles-sink.c $TERMUX_PKG_SRCDIR/src/modules/sles
+	mkdir $TERMUX_PKG_SRCDIR/src/modules/aaudio
+	cp $TERMUX_PKG_BUILDER_DIR/module-aaudio-sink.c $TERMUX_PKG_SRCDIR/src/modules/aaudio
 	intltoolize --automake --copy --force
 	LDFLAGS+=" -llog -landroid-glob"
 }
@@ -39,6 +41,7 @@ termux_step_post_make_install () {
 	sed -i $TERMUX_PREFIX/etc/pulse/default.pa \
 		-e '/^load-module module-detect$/s/^/#/'
 	echo "load-module module-sles-sink" >> $TERMUX_PREFIX/etc/pulse/default.pa
+	echo "#load-module module-aaudio-sink" >> $TERMUX_PREFIX/etc/pulse/default.pa
 
 	if [ "$TERMUX_ARCH_BITS" = 32 ]; then
 		SYSTEM_LIB=lib
diff --git a/packages/libpulseaudio/makefile.am.patch b/packages/libpulseaudio/makefile.am.patch
deleted file mode 100644
index edafaf7a0..000000000
--- a/packages/libpulseaudio/makefile.am.patch
+++ /dev/null
@@ -1,14 +0,0 @@
---- ../cache/pulseaudio-11.1/src/Makefile.am	2017-09-18 10:41:02.000000000 +0000
-+++ ./src/Makefile.am	2017-12-22 22:51:30.628573929 +0000
-@@ -1594,6 +1595,11 @@
- module_simple_protocol_unix_la_LIBADD = $(MODULE_LIBADD) libprotocol-simple.la
- 
- # CLI protocol
-+modlibexec_LTLIBRARIES += module-sles-sink.la
-+module_sles_sink_la_SOURCES = modules/sles/module-sles-sink.c
-+module_sles_sink_la_LDFLAGS = $(MODULE_LDFLAGS) -lOpenSLES
-+module_sles_sink_la_LIBADD = $(MODULE_LIBADD)
-+module_sles_sink_la_CFLAGS = $(AM_CFLAGS) -DPA_MODULE_NAME=module_sles_sink
- 
- module_cli_la_SOURCES = modules/module-cli.c
- module_cli_la_LDFLAGS = $(MODULE_LDFLAGS)
diff --git a/packages/libpulseaudio/module-aaudio-sink.c b/packages/libpulseaudio/module-aaudio-sink.c
new file mode 100644
index 000000000..f67b40d24
--- /dev/null
+++ b/packages/libpulseaudio/module-aaudio-sink.c
@@ -0,0 +1,401 @@
+/***
+  This file is part of PulseAudio.
+
+  Copyright 2004-2008 Lennart Poettering
+
+  PulseAudio is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser General Public License as published
+  by the Free Software Foundation; either version 2.1 of the License,
+  or (at your option) any later version.
+
+  PulseAudio is distributed in the hope that it will be useful, but
+  WITHOUT ANY WARRANTY; without even the implied warranty of
+  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+  General Public License for more details.
+
+  You should have received a copy of the GNU Lesser General Public License
+  along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <errno.h>
+#include <unistd.h>
+
+#include <pulse/rtclock.h>
+#include <pulse/timeval.h>
+#include <pulse/xmalloc.h>
+
+#include <pulsecore/i18n.h>
+#include <pulsecore/macro.h>
+#include <pulsecore/sink.h>
+#include <pulsecore/module.h>
+#include <pulsecore/core-util.h>
+#include <pulsecore/modargs.h>
+#include <pulsecore/log.h>
+#include <pulsecore/thread.h>
+#include <pulsecore/thread-mq.h>
+#include <pulsecore/rtpoll.h>
+
+#include <aaudio/AAudio.h>
+
+PA_MODULE_AUTHOR("Tom Yan");
+PA_MODULE_DESCRIPTION("Android AAudio sink");
+PA_MODULE_VERSION(PACKAGE_VERSION);
+PA_MODULE_LOAD_ONCE(false);
+PA_MODULE_USAGE(
+    "sink_name=<name for the sink> "
+    "sink_properties=<properties for the sink> "
+    "rate=<sampling rate> "
+    "latency=<buffer length> "
+    "no_close_hack=<avoid segfault caused by AAudioStream_close()> "
+);
+
+#define DEFAULT_SINK_NAME "AAudio sink"
+
+struct userdata {
+    pa_core *core;
+    pa_module *module;
+    pa_sink *sink;
+
+    pa_thread *thread;
+    pa_thread_mq thread_mq;
+    pa_rtpoll *rtpoll;
+
+    uint32_t latency;
+    uint32_t rate;
+    bool no_close;
+
+    pa_memchunk memchunk;
+    size_t frame_size;
+
+    AAudioStreamBuilder *builder;
+    AAudioStream *stream;
+    pa_sample_spec ss;
+};
+
+static const char* const valid_modargs[] = {
+    "sink_name",
+    "sink_properties",
+    "rate",
+    "latency",
+    "no_close_hack",
+    NULL
+};
+
+static void update_latency(struct userdata *u) {
+    pa_usec_t block_usec;
+
+    if(!u->latency) {
+        block_usec = PA_USEC_PER_SEC * AAudioStream_getBufferSizeInFrames(u->stream) / u->sink->sample_spec.rate / 2;
+        if(!pa_thread_mq_get())
+            pa_sink_set_fixed_latency(u->sink, block_usec);
+        else
+            pa_sink_set_fixed_latency_within_thread(u->sink, block_usec);
+    }
+}
+
+static aaudio_data_callback_result_t buffer_callback(AAudioStream *stream, void *userdata, void *audioData, int32_t numFrames) {
+    struct userdata* u = userdata;
+
+    pa_assert(u);
+
+    if (!pa_thread_mq_get()) {
+        pa_log_debug("Thread starting up");
+        pa_thread_mq_install(&u->thread_mq);
+    }
+
+    u->memchunk.memblock = pa_memblock_new_fixed(u->core->mempool, audioData, numFrames * u->frame_size, false);
+    u->memchunk.length = numFrames * u->frame_size;
+    pa_sink_render_into_full(u->sink, &u->memchunk);
+    pa_memblock_unref_fixed(u->memchunk.memblock);
+
+    return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+static void error_callback(AAudioStream *stream, void *userdata, aaudio_result_t error) {
+    struct userdata* u = userdata;
+
+    pa_assert(u);
+
+    if (error == AAUDIO_ERROR_DISCONNECTED) {
+        if (!pa_thread_mq_get()) {
+            pa_sink_suspend(u->sink, true, PA_SUSPEND_UNAVAILABLE);
+            pa_sink_suspend(u->sink, false, PA_SUSPEND_UNAVAILABLE);
+        } else {
+            AAudioStream_requestStop(u->stream);
+            AAudioStream_requestStart(u->stream);
+            update_latency(u);
+            pa_log("Failed to reconfigure sink for new device.\n");
+        }
+    }
+}
+
+#define CHK(stmt) { \
+    aaudio_result_t res = stmt; \
+    if (res != AAUDIO_OK) { \
+        fprintf(stderr, "error %d at %s:%d\n", res, __FILE__, __LINE__); \
+        goto fail; \
+    } \
+}
+
+static int pa_open_aaudio_stream(struct userdata *u)
+{
+    bool want_float;
+    aaudio_format_t format;
+    pa_sample_spec *ss = &u->ss;
+
+    CHK(AAudio_createStreamBuilder(&u->builder));
+    AAudioStreamBuilder_setPerformanceMode(u->builder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+    AAudioStreamBuilder_setDataCallback(u->builder, buffer_callback, u);
+    AAudioStreamBuilder_setErrorCallback(u->builder, error_callback, u);
+
+    want_float = ss->format > PA_SAMPLE_S16BE;
+    ss->format = want_float ? PA_SAMPLE_FLOAT32LE : PA_SAMPLE_S16LE;
+    format = want_float ? AAUDIO_FORMAT_PCM_FLOAT : AAUDIO_FORMAT_PCM_I16;
+    AAudioStreamBuilder_setFormat(u->builder, format);
+
+    if (u->rate)
+        AAudioStreamBuilder_setSampleRate(u->builder, u->rate);
+
+    AAudioStreamBuilder_setChannelCount(u->builder, ss->channels);
+
+    CHK(AAudioStreamBuilder_openStream(u->builder, &u->stream));
+    CHK(AAudioStreamBuilder_delete(u->builder));
+
+    ss->rate = AAudioStream_getSampleRate(u->stream);
+    u->frame_size = pa_frame_size(ss);
+
+    return 0;
+
+fail:
+    return -1;
+}
+
+#undef CHK
+
+static void thread_func(void *userdata) {
+    struct userdata *u = userdata;
+
+    pa_assert(u);
+
+    pa_log_debug("Thread starting up");
+    pa_thread_mq_install(&u->thread_mq);
+
+    for (;;) {
+        int ret;
+
+        if (PA_SINK_IS_LINKED(u->sink->thread_info.state)) {
+            AAudioStream_requestStart(u->stream);
+            update_latency(u);
+            break;
+        }
+
+        /* Hmm, nothing to do. Let's sleep */
+        if ((ret = pa_rtpoll_run(u->rtpoll)) < 0)
+            goto fail;
+
+        if (ret == 0)
+            goto finish;
+    }
+
+    for (;;) {
+        int ret;
+
+        /* Hmm, nothing to do. Let's sleep */
+        if ((ret = pa_rtpoll_run(u->rtpoll)) < 0)
+            goto fail;
+
+        if (ret == 0)
+            goto finish;
+    }
+
+fail:
+    /* If this was no regular exit from the loop we have to continue
+     * processing messages until we received PA_MESSAGE_SHUTDOWN */
+    pa_asyncmsgq_post(u->thread_mq.outq, PA_MSGOBJECT(u->core), PA_CORE_MESSAGE_UNLOAD_MODULE, u->module, 0, NULL, NULL);
+    pa_asyncmsgq_wait_for(u->thread_mq.inq, PA_MESSAGE_SHUTDOWN);
+
+finish:
+    pa_log_debug("Thread shutting down");
+}
+
+static int state_func(pa_sink *s, pa_sink_state_t state, pa_suspend_cause_t suspend_cause) {
+    struct userdata *u = s->userdata;
+    int r = 0;
+    uint32_t idx;
+    pa_sink_input *i;
+    pa_idxset *inputs;
+
+    if ((PA_SINK_IS_OPENED(s->state) && state == PA_SINK_SUSPENDED) ||
+        (PA_SINK_IS_LINKED(s->state) && state == PA_SINK_UNLINKED)) {
+        if (u->no_close) {
+            AAudioStream_requestStop(u->stream);
+        } else {
+            AAudioStream_close(u->stream);
+        }
+    } else if (s->state == PA_SINK_SUSPENDED && PA_SINK_IS_OPENED(state)) {
+        pa_open_aaudio_stream(u);
+
+        inputs = pa_idxset_copy(s->inputs, NULL);
+        PA_IDXSET_FOREACH(i, inputs, idx) {
+            if (i->state == PA_SINK_INPUT_RUNNING) {
+                pa_sink_input_cork(i, true);
+            } else {
+                pa_idxset_remove_by_index(inputs, idx);
+            }
+        }
+
+        s->alternate_sample_rate = u->ss.rate;
+        pa_sink_reconfigure(s, &u->ss, false);
+        s->default_sample_rate = u->ss.rate;
+
+        /* Avoid infinite loop triggered if uncork in this case */
+        if (s->suspend_cause == PA_SUSPEND_IDLE)
+            pa_sink_suspend(u->sink, true, PA_SUSPEND_UNAVAILABLE);
+
+        PA_IDXSET_FOREACH(i, inputs, idx) pa_sink_input_cork(i, false);
+        pa_idxset_free(inputs, NULL);
+
+        AAudioStream_requestStart(u->stream);
+        update_latency(u);
+    }
+    return r;
+}
+
+static int reconfigure_func(pa_sink *s, pa_sample_spec *ss, bool passthrough) {
+    s->sample_spec.rate = ss->rate;
+    return 0;
+}
+
+static void process_rewind(pa_sink *s) {
+    pa_sink_process_rewind(s, 0);
+}
+
+int pa__init(pa_module*m) {
+    struct userdata *u = NULL;
+    pa_channel_map map;
+    pa_modargs *ma = NULL;
+    pa_sink_new_data data;
+    pa_usec_t block_usec;
+
+    pa_assert(m);
+
+    m->userdata = u = pa_xnew0(struct userdata, 1);
+
+    u->core = m->core;
+    u->module = m;
+    u->rtpoll = pa_rtpoll_new();
+    pa_thread_mq_init(&u->thread_mq, m->core->mainloop, u->rtpoll);
+
+    if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
+        pa_log("Failed to parse module arguments.");
+        goto fail;
+    }
+
+    u->ss = m->core->default_sample_spec;
+    map = m->core->default_channel_map;
+    pa_modargs_get_sample_rate(ma, &u->rate);
+
+    pa_modargs_get_value_boolean(ma, "no_close_hack", &u->no_close);
+
+    if (pa_open_aaudio_stream(u) < 0)
+        goto fail;
+
+    pa_sink_new_data_init(&data);
+    data.driver = __FILE__;
+    data.module = m;
+    pa_sink_new_data_set_name(&data, pa_modargs_get_value(ma, "sink_name", DEFAULT_SINK_NAME));
+    pa_sink_new_data_set_sample_spec(&data, &u->ss);
+    pa_sink_new_data_set_channel_map(&data, &map);
+    pa_proplist_sets(data.proplist, PA_PROP_DEVICE_DESCRIPTION, _("AAudio Output"));
+    pa_proplist_sets(data.proplist, PA_PROP_DEVICE_CLASS, "abstract");
+
+    if (pa_modargs_get_proplist(ma, "sink_properties", data.proplist, PA_UPDATE_REPLACE) < 0) {
+        pa_log("Invalid properties");
+        pa_sink_new_data_done(&data);
+        goto fail;
+    }
+
+    u->sink = pa_sink_new(m->core, &data, 0);
+    pa_sink_new_data_done(&data);
+
+    if (!u->sink) {
+        pa_log("Failed to create sink object.");
+        goto fail;
+    }
+
+    u->sink->parent.process_msg = pa_sink_process_msg;
+    u->sink->set_state_in_main_thread = state_func;
+    u->sink->reconfigure = reconfigure_func;
+    u->sink->request_rewind = process_rewind;
+    u->sink->userdata = u;
+
+    pa_sink_set_asyncmsgq(u->sink, u->thread_mq.inq);
+    pa_sink_set_rtpoll(u->sink, u->rtpoll);
+
+    pa_modargs_get_value_u32(ma, "latency", &u->latency);
+    if (u->latency) {
+        block_usec = PA_USEC_PER_MSEC * u->latency;
+        pa_sink_set_fixed_latency(u->sink, block_usec);
+    }
+
+    if (!(u->thread = pa_thread_new("aaudio-sink", thread_func, u))) {
+        pa_log("Failed to create thread.");
+        goto fail;
+    }
+
+    pa_sink_put(u->sink);
+
+    pa_modargs_free(ma);
+
+    return 0;
+
+fail:
+    if (ma)
+        pa_modargs_free(ma);
+
+    pa__done(m);
+
+    return -1;
+}
+
+int pa__get_n_used(pa_module *m) {
+    struct userdata *u;
+
+    pa_assert(m);
+    pa_assert_se(u = m->userdata);
+
+    return pa_sink_linked_by(u->sink);
+}
+
+void pa__done(pa_module*m) {
+    struct userdata *u;
+
+    pa_assert(m);
+
+    if (!(u = m->userdata))
+        return;
+
+    if (u->sink)
+        pa_sink_unlink(u->sink);
+
+    if (u->thread) {
+        pa_asyncmsgq_send(u->thread_mq.inq, NULL, PA_MESSAGE_SHUTDOWN, NULL, 0, NULL);
+        pa_thread_free(u->thread);
+    }
+
+    pa_thread_mq_done(&u->thread_mq);
+
+    if (u->sink)
+        pa_sink_unref(u->sink);
+
+    if (u->rtpoll)
+        pa_rtpoll_free(u->rtpoll);
+
+    pa_xfree(u);
+}
diff --git a/packages/libpulseaudio/sles.patch b/packages/libpulseaudio/sles.patch
new file mode 100644
index 000000000..98c26f539
--- /dev/null
+++ b/packages/libpulseaudio/sles.patch
@@ -0,0 +1,17 @@
+diff --git a/src/Makefile.am~ b/src/Makefile.am
+index f4464d2..a2c201d 100644
+--- a/src/Makefile.am~
++++ b/src/Makefile.am
+@@ -1495,6 +1495,12 @@ bin_SCRIPTS += utils/qpaeq
+ endif
+ endif
+ 
++modlibexec_LTLIBRARIES += module-sles-sink.la
++module_sles_sink_la_SOURCES = modules/sles/module-sles-sink.c
++module_sles_sink_la_LDFLAGS = $(MODULE_LDFLAGS) -lOpenSLES
++module_sles_sink_la_LIBADD = $(MODULE_LIBADD)
++module_sles_sink_la_CFLAGS = $(AM_CFLAGS) -DPA_MODULE_NAME=module_sles_sink
++
+ # Simple protocol
+ 
+ module_simple_protocol_tcp_la_SOURCES = modules/module-protocol-stub.c