diff --git a/pulseaudio-module/CMakeLists.txt b/pulseaudio-module/CMakeLists.txt
index f7a1ff4..5beebbd 100644
--- a/pulseaudio-module/CMakeLists.txt
+++ b/pulseaudio-module/CMakeLists.txt
@@ -26,3 +26,17 @@ target_link_libraries(module-aaudio-sink
pulsecommon-13.0
pulse
aaudio)
+
+add_library(module-aaudio-source SHARED
+ module-aaudio-source.c)
+
+set_target_properties(module-aaudio-source PROPERTIES
+ OUTPUT_NAME "module-aaudio-source"
+ PREFIX ""
+ SUFFIX ".so")
+
+target_link_libraries(module-aaudio-source
+ pulsecore-13.0
+ pulsecommon-13.0
+ pulse
+ aaudio)
diff --git a/pulseaudio-module/build.sh b/pulseaudio-module/build.sh
index a76c69c..c947843 100755
--- a/pulseaudio-module/build.sh
+++ b/pulseaudio-module/build.sh
@@ -15,9 +15,11 @@ export CFLAGS="-I../pulseaudio/build-arm64 -I../pulseaudio/src -I../root-arm64/i
export LDFLAGS="-L../root-arm64/lib/pulseaudio -L../root-arm64/lib -Wl,-z,max-page-size=16384 -Wl,-z,common-page-size=16384"
$CC -O2 -shared $CFLAGS $LDFLAGS -lpulsecore-13.0 -lpulsecommon-13.0 -lpulse -laaudio -llog -o build64/module-aaudio-sink.so module-aaudio-sink.c
+$CC -O2 -shared $CFLAGS $LDFLAGS -lpulsecore-13.0 -lpulsecommon-13.0 -lpulse -laaudio -llog -o build64/module-aaudio-source.so module-aaudio-source.c
export CC="$TOOLCHAIN/armv7a-linux-androideabi26-clang"
export CFLAGS="-I../pulseaudio/build-armhf -I../pulseaudio/src -I../root-armhf/include"
export LDFLAGS="-L../root-armhf/lib/pulseaudio -L../root-armhf/lib -Wl,-z,max-page-size=16384 -Wl,-z,common-page-size=16384"
-$CC -O2 -shared $CFLAGS $LDFLAGS -lpulsecore-13.0 -lpulsecommon-13.0 -lpulse -laaudio -llog -o build/module-aaudio-sink.so module-aaudio-sink.c
\ No newline at end of file
+$CC -O2 -shared $CFLAGS $LDFLAGS -lpulsecore-13.0 -lpulsecommon-13.0 -lpulse -laaudio -llog -o build/module-aaudio-sink.so module-aaudio-sink.c
+$CC -O2 -shared $CFLAGS $LDFLAGS -lpulsecore-13.0 -lpulsecommon-13.0 -lpulse -laaudio -llog -o build/module-aaudio-source.so module-aaudio-source.c
\ No newline at end of file
diff --git a/pulseaudio-module/create-asset.sh b/pulseaudio-module/create-asset.sh
index 2d3ea18..c48641b 100755
--- a/pulseaudio-module/create-asset.sh
+++ b/pulseaudio-module/create-asset.sh
@@ -7,6 +7,7 @@ cp -a ../output/aarch64-linux-gnu/modules/libprotocol-native.so pulseaudio/modul
cp -a ../output/aarch64-linux-gnu/modules/module-native-protocol-unix.so pulseaudio/modules
cp -a build64/module-aaudio-sink.so pulseaudio/modules
+cp -a build64/module-aaudio-source.so pulseaudio/modules
tar -I 'zstd --ultra -22' -cf pulseaudio-gamenative.tzst -C pulseaudio .
diff --git a/pulseaudio-module/module-aaudio-source.c b/pulseaudio-module/module-aaudio-source.c
new file mode 100644
index 0000000..7285eec
--- /dev/null
+++ b/pulseaudio-module/module-aaudio-source.c
@@ -0,0 +1,596 @@
+/***
+ 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 .
+***/
+
+/* AAudio capture source, the input counterpart to module-aaudio-sink.
+ *
+ * Without a real source, the only capture device PulseAudio offers is
+ * AAudioSink.monitor. Wine enumerates monitors as Windows recording devices, so
+ * games pick it up as a microphone and end up transmitting their own output -
+ * in a voice chat that means rebroadcasting the whole lobby back into it.
+ *
+ * Structure deliberately mirrors module-aaudio-sink.c: an AAudio callback hands
+ * buffers to the PulseAudio I/O thread over an asyncmsgq, the same recreate /
+ * try-start / try-stop message pattern handles stream loss, and the state machine
+ * lives in set_state_in_io_thread.
+ *
+ * Two deliberate differences from the sink:
+ *
+ * - The AAudio stream is stopped whenever the source is not RUNNING, not just
+ * when it is suspended. A sink holding an idle output stream costs nothing,
+ * but a source holding an idle input stream keeps the microphone open, which
+ * lights the Android privacy indicator for the whole session and runs into
+ * the background-capture restrictions on Android 14+.
+ *
+ * - pa__done stops the stream and joins the I/O thread before returning, so
+ * unloading the module actually releases the microphone.
+ */
+
+#include
+
+#include
+
+#include
+#include
+#include
+#include
+#include
+
+#include
+#undef __INTRODUCED_IN
+#define __INTRODUCED_IN(api_level)
+#include
+#include
+
+#define LOG_TAG "GN-PulseAudioSource"
+#define LOGW(...) __android_log_print(ANDROID_LOG_WARN, LOG_TAG, __VA_ARGS__)
+
+PA_MODULE_AUTHOR("Tom Yan, BrunoSX, Joshua Tam");
+PA_MODULE_DESCRIPTION("Winlator AAudio source");
+PA_MODULE_VERSION(PACKAGE_VERSION);
+PA_MODULE_LOAD_ONCE(false);
+PA_MODULE_USAGE(
+ "source_name= "
+ "source_properties= "
+ "rate= "
+ "channels= "
+ "performance_mode= "
+ "input_preset= "
+ "set_default="
+);
+
+#define DEFAULT_SOURCE_NAME "AAudioSource"
+
+/* AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION - gives us the hardware echo canceller
+ * and noise suppressor, which matters because the game's own output is usually
+ * coming out of the same device's speaker. */
+#define DEFAULT_INPUT_PRESET 7
+
+enum {
+ SOURCE_MESSAGE_CAPTURE = PA_SOURCE_MESSAGE_MAX,
+ SOURCE_MESSAGE_TRY_START,
+ SOURCE_MESSAGE_TRY_STOP,
+ SOURCE_MESSAGE_RECREATE,
+};
+
+struct userdata {
+ pa_core *core;
+ pa_module *module;
+ pa_source *source;
+
+ pa_thread *thread;
+ pa_thread_mq thread_mq;
+ pa_rtpoll *rtpoll;
+ pa_rtpoll_item *rtpoll_item;
+ pa_asyncmsgq *aaudio_msgq;
+
+ size_t frame_size;
+
+ AAudioStreamBuilder *builder;
+ AAudioStream *stream;
+ pa_sample_spec ss;
+
+ int performance_mode;
+ int input_preset;
+ bool set_default;
+
+ int32_t frames_per_burst;
+ int32_t device_sample_rate;
+ int32_t device_channels;
+ int32_t last_device_id;
+
+ bool stream_started;
+};
+
+static const char* const valid_modargs[] = {
+ "source_name",
+ "source_properties",
+ "rate",
+ "channels",
+ "performance_mode",
+ "input_preset",
+ "set_default",
+ NULL
+};
+
+/* AAudioStreamBuilder_setInputPreset is API 28, and this module is built at API 26.
+ * Blanking __INTRODUCED_IN above lets it compile, but it also strips the annotation that
+ * would otherwise have made the reference weak, leaving a strong undefined symbol - and
+ * the module is linked with -z now, so the dynamic linker resolves every undefined symbol
+ * when the module is dlopen'd, before any of our code runs. On an API 26/27 device
+ * libaaudio.so does not export it, so the whole module would fail to load, and a runtime
+ * API-level check could not prevent that because it never gets the chance to run.
+ * Declaring it weak lets the linker leave the address NULL instead of failing the load,
+ * which is what makes the check below meaningful. Same approach as the sink. */
+extern __attribute__((weak)) void AAudioStreamBuilder_setInputPreset(AAudioStreamBuilder *builder,
+ aaudio_input_preset_t preset);
+
+static void try_set_input_preset(AAudioStreamBuilder *builder, int preset) {
+ static bool warned = false;
+
+ if (AAudioStreamBuilder_setInputPreset) {
+ AAudioStreamBuilder_setInputPreset(builder, preset);
+ return;
+ }
+
+ if (!warned) {
+ LOGW("AAudioStreamBuilder_setInputPreset unavailable (API < 28), using the default input preset");
+ warned = true;
+ }
+}
+
+static void schedule_start(struct userdata *u) {
+ pa_asyncmsgq_post(u->thread_mq.inq, PA_MSGOBJECT(u->source), SOURCE_MESSAGE_TRY_START, NULL, 0, NULL, NULL);
+}
+
+static void schedule_stop(struct userdata *u) {
+ pa_asyncmsgq_post(u->thread_mq.inq, PA_MSGOBJECT(u->source), SOURCE_MESSAGE_TRY_STOP, NULL, 0, NULL, NULL);
+}
+
+static void schedule_recreate(struct userdata *u) {
+ pa_asyncmsgq_post(u->thread_mq.inq, PA_MSGOBJECT(u->source), SOURCE_MESSAGE_RECREATE, NULL, 0, NULL, NULL);
+}
+
+/* Runs on the AAudio callback thread. Hand the buffer to the I/O thread and block
+ * until it has been posted, so audioData stays valid for the whole call. */
+static aaudio_data_callback_result_t aaudio_data_callback(AAudioStream *stream, void *userdata, void *audioData, int32_t numFrames) {
+ struct userdata *u = userdata;
+ return pa_asyncmsgq_send(u->aaudio_msgq, PA_MSGOBJECT(u->source), SOURCE_MESSAGE_CAPTURE, audioData, numFrames, NULL);
+}
+
+static void aaudio_error_callback(AAudioStream *stream, void *userdata, aaudio_result_t error) {
+ struct userdata *u = userdata;
+
+ if (error == AAUDIO_ERROR_DISCONNECTED ||
+ error == AAUDIO_ERROR_INVALID_STATE ||
+ error == AAUDIO_ERROR_INVALID_HANDLE ||
+ error == AAUDIO_ERROR_TIMEOUT) {
+ LOGW("AAudio stream error (%d), attempting to reconnect...", error);
+ schedule_recreate(u);
+ } else {
+ LOGW("AAudio error callback: %d", error);
+ }
+}
+
+static pa_usec_t get_aaudio_latency(struct userdata *u) {
+ if (u->stream != NULL && u->ss.rate > 0) {
+ int32_t buffer_size = AAudioStream_getBufferSizeInFrames(u->stream);
+ if (buffer_size > 0) {
+ return PA_USEC_PER_SEC * (int64_t) (buffer_size + u->frames_per_burst) / u->ss.rate;
+ }
+ }
+
+ return 20 * PA_USEC_PER_MSEC;
+}
+
+static void update_pa_latency(struct userdata *u) {
+ if (u->source) {
+ pa_usec_t latency = get_aaudio_latency(u);
+ if (pa_thread_mq_get()) {
+ pa_source_set_fixed_latency_within_thread(u->source, latency);
+ } else {
+ pa_source_set_fixed_latency(u->source, latency);
+ }
+ }
+}
+
+static int pa_create_aaudio_stream(struct userdata *u) {
+ aaudio_result_t res;
+
+ res = AAudio_createStreamBuilder(&u->builder);
+ if (res != AAUDIO_OK) {
+ LOGW("AAudio_createStreamBuilder() failed.");
+ return -1;
+ }
+
+ AAudioStreamBuilder_setDirection(u->builder, AAUDIO_DIRECTION_INPUT);
+ AAudioStreamBuilder_setPerformanceMode(u->builder, u->performance_mode);
+ AAudioStreamBuilder_setDataCallback(u->builder, aaudio_data_callback, u);
+ AAudioStreamBuilder_setErrorCallback(u->builder, aaudio_error_callback, u);
+ AAudioStreamBuilder_setFormat(u->builder, u->ss.format == PA_SAMPLE_FLOAT32LE ? AAUDIO_FORMAT_PCM_FLOAT : AAUDIO_FORMAT_PCM_I16);
+
+ /* Let AAudio pick the device's native capture rate and channel count and adapt
+ * the source to whatever we are given, rather than forcing a resample here. */
+ AAudioStreamBuilder_setSampleRate(u->builder, AAUDIO_UNSPECIFIED);
+ AAudioStreamBuilder_setChannelCount(u->builder, AAUDIO_UNSPECIFIED);
+
+ try_set_input_preset(u->builder, u->input_preset);
+
+ res = AAudioStreamBuilder_openStream(u->builder, &u->stream);
+ if (res != AAUDIO_OK) {
+ /* The most likely cause here is a missing or revoked RECORD_AUDIO grant. */
+ LOGW("AAudioStreamBuilder_openStream() failed: %d (%s)", res, AAudio_convertResultToText(res));
+ AAudioStreamBuilder_delete(u->builder);
+ u->builder = NULL;
+ return -1;
+ }
+
+ AAudioStreamBuilder_delete(u->builder);
+ u->builder = NULL;
+
+ u->device_sample_rate = AAudioStream_getSampleRate(u->stream);
+ u->device_channels = AAudioStream_getChannelCount(u->stream);
+ aaudio_format_t actual_format = AAudioStream_getFormat(u->stream);
+
+ LOGW("AAudio input stream opened: %d Hz, %d channels, format %d, device %d",
+ u->device_sample_rate, u->device_channels, actual_format,
+ AAudioStream_getDeviceId(u->stream));
+
+ u->ss.rate = u->device_sample_rate;
+ u->ss.channels = u->device_channels;
+
+ if (actual_format == AAUDIO_FORMAT_PCM_FLOAT) {
+ u->ss.format = PA_SAMPLE_FLOAT32LE;
+ } else if (actual_format == AAUDIO_FORMAT_PCM_I16) {
+ u->ss.format = PA_SAMPLE_S16LE;
+ }
+
+ u->frames_per_burst = AAudioStream_getFramesPerBurst(u->stream);
+ u->last_device_id = AAudioStream_getDeviceId(u->stream);
+ u->frame_size = pa_frame_size(&u->ss);
+ u->stream_started = false;
+
+ update_pa_latency(u);
+
+ return 0;
+}
+
+static void try_start_stream(struct userdata *u) {
+ aaudio_result_t res;
+
+ if (!u->stream || u->stream_started) return;
+
+ res = AAudioStream_requestStart(u->stream);
+ if (res == AAUDIO_OK) {
+ u->stream_started = true;
+ LOGW("AAudioStream_requestStart() succeeded");
+ } else {
+ LOGW("AAudioStream_requestStart() failed: %d, scheduling recreate", res);
+ schedule_recreate(u);
+ }
+}
+
+static void try_stop_stream(struct userdata *u) {
+ aaudio_result_t res;
+
+ if (!u->stream || !u->stream_started) return;
+
+ res = AAudioStream_requestStop(u->stream);
+ if (res != AAUDIO_OK) {
+ LOGW("AAudioStream_requestStop() failed: %d", res);
+ } else {
+ LOGW("AAudio input stream stopped, microphone released");
+ }
+ u->stream_started = false;
+}
+
+static void recreate_aaudio_stream(struct userdata *u) {
+ if (u->stream) {
+ AAudioStream_requestStop(u->stream);
+ AAudioStream_close(u->stream);
+ u->stream = NULL;
+ u->stream_started = false;
+ }
+
+ if (pa_create_aaudio_stream(u) < 0) {
+ LOGW("Failed to create AAudio input stream");
+ /* Deliberately not rescheduling: if the microphone permission is denied,
+ * retrying forever would spin. The next state transition will retry. */
+ } else {
+ LOGW("AAudio input stream created, attempting to start");
+ schedule_start(u);
+ }
+}
+
+static int source_process_capture(struct userdata *u, void *audioData, int64_t numFrames) {
+ pa_memchunk chunk;
+
+ if (!PA_SOURCE_IS_LINKED(u->source->thread_info.state)) return AAUDIO_CALLBACK_RESULT_STOP;
+
+ /* Drop the buffer rather than posting it while suspended or not running. */
+ if (u->source->thread_info.state != PA_SOURCE_RUNNING) return AAUDIO_CALLBACK_RESULT_CONTINUE;
+
+ if (numFrames <= 0) return AAUDIO_CALLBACK_RESULT_CONTINUE;
+
+ /* audioData belongs to AAudio and is only valid for this call. pa_source_post
+ * copies into the source outputs' buffers before returning, and the callback
+ * thread is blocked in pa_asyncmsgq_send until we get here, so wrapping it
+ * read-only without copying is safe. */
+ chunk.memblock = pa_memblock_new_fixed(u->core->mempool, audioData, u->frame_size * numFrames, true);
+ chunk.index = 0;
+ chunk.length = u->frame_size * numFrames;
+
+ pa_source_post(u->source, &chunk);
+
+ pa_memblock_unref_fixed(chunk.memblock);
+
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+static int source_process_msg(pa_msgobject *o, int code, void *data, int64_t offset, pa_memchunk *memchunk) {
+ struct userdata *u = PA_SOURCE(o)->userdata;
+
+ if (code == SOURCE_MESSAGE_CAPTURE) return source_process_capture(u, data, offset);
+
+ if (code == SOURCE_MESSAGE_TRY_START) {
+ LOGW("AAudio try start requested");
+ if (u->source->thread_info.state == PA_SOURCE_RUNNING) {
+ try_start_stream(u);
+ } else {
+ LOGW("Source no longer running, canceling try");
+ }
+ return 0;
+ }
+
+ if (code == SOURCE_MESSAGE_TRY_STOP) {
+ LOGW("AAudio try stop requested");
+ try_stop_stream(u);
+ return 0;
+ }
+
+ if (code == SOURCE_MESSAGE_RECREATE) {
+ LOGW("AAudio recreate requested");
+ if (PA_SOURCE_IS_LINKED(u->source->thread_info.state)) {
+ recreate_aaudio_stream(u);
+ } else {
+ LOGW("Source no longer linked, canceling recreate");
+ }
+ return 0;
+ }
+
+ return pa_source_process_msg(o, code, data, offset, memchunk);
+}
+
+static int source_set_state_io_thread(pa_source *s, pa_source_state_t state, pa_suspend_cause_t suspend_cause) {
+ struct userdata *u = s->userdata;
+
+ LOGW("AAudio source state transition: current=%d, target=%d, suspend_cause=%d",
+ s->thread_info.state, state, suspend_cause);
+
+ /* Hold the microphone open only while something is actually recording. */
+ if (state == PA_SOURCE_RUNNING) {
+ if (!u->stream) {
+ LOGW("No AAudio stream, recreating before start");
+ recreate_aaudio_stream(u);
+ } else {
+ aaudio_stream_state_t stream_state = AAudioStream_getState(u->stream);
+ if (stream_state == AAUDIO_STREAM_STATE_STARTED ||
+ stream_state == AAUDIO_STREAM_STATE_STARTING) {
+ u->stream_started = true;
+ } else if (stream_state == AAUDIO_STREAM_STATE_DISCONNECTED) {
+ LOGW("AAudio stream disconnected, recreating");
+ schedule_recreate(u);
+ } else {
+ schedule_start(u);
+ }
+ }
+ } else {
+ /* IDLE, SUSPENDED or UNLINKED - all mean nobody is recording right now. */
+ if (u->stream_started) {
+ schedule_stop(u);
+ }
+ }
+
+ return 0;
+}
+
+static void thread_func(void *userdata) {
+ struct userdata *u = userdata;
+ pa_thread_mq_install(&u->thread_mq);
+
+ for (;;) {
+ int res = pa_rtpoll_run(u->rtpoll);
+ if (res < 0) {
+ goto error;
+ } else if (res == 0) {
+ break;
+ }
+ }
+
+ return;
+
+error:
+ 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);
+}
+
+void pa__done(pa_module *m) {
+ struct userdata *u;
+
+ if (!(u = m->userdata)) return;
+
+ if (u->source) pa_source_unlink(u->source);
+
+ /* Join the I/O thread before touching the stream, so nothing is mid-post. */
+ 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->source) pa_source_unref(u->source);
+
+ /* Stop before close so the microphone is released promptly and the Android
+ * privacy indicator clears. */
+ if (u->stream) {
+ AAudioStream_requestStop(u->stream);
+ AAudioStream_close(u->stream);
+ u->stream = NULL;
+ }
+
+ if (u->builder) AAudioStreamBuilder_delete(u->builder);
+ if (u->rtpoll_item) pa_rtpoll_item_free(u->rtpoll_item);
+ if (u->aaudio_msgq) pa_asyncmsgq_unref(u->aaudio_msgq);
+ if (u->rtpoll) pa_rtpoll_free(u->rtpoll);
+
+ pa_xfree(u);
+}
+
+int pa__init(pa_module *m) {
+ struct userdata *u = NULL;
+ pa_modargs *ma = NULL;
+ pa_channel_map map;
+ pa_source_new_data data;
+ int performance_mode = 0;
+
+ if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
+ LOGW("Failed to parse module arguments.");
+ goto error;
+ }
+
+ m->userdata = u = pa_xnew0(struct userdata, 1);
+
+ u->core = m->core;
+ u->module = m;
+ u->rtpoll = pa_rtpoll_new();
+
+ if (pa_thread_mq_init(&u->thread_mq, m->core->mainloop, u->rtpoll) < 0) {
+ LOGW("pa_thread_mq_init() failed.");
+ goto error;
+ }
+
+ u->aaudio_msgq = pa_asyncmsgq_new(0);
+ if (!u->aaudio_msgq) {
+ LOGW("pa_asyncmsgq_new() failed.");
+ goto error;
+ }
+
+ u->rtpoll_item = pa_rtpoll_item_new_asyncmsgq_read(u->rtpoll, PA_RTPOLL_EARLY-1, u->aaudio_msgq);
+
+ u->ss = m->core->default_sample_spec;
+ map = m->core->default_channel_map;
+
+ if (pa_modargs_get_sample_spec_and_channel_map(ma, &u->ss, &map, PA_CHANNEL_MAP_DEFAULT) < 0) {
+ LOGW("pa_modargs_get_sample_spec_and_channel_map() failed.");
+ goto error;
+ }
+
+ /* Mono is what voice chat wants; overridden below by whatever AAudio opens. */
+ u->ss.channels = 1;
+ u->ss.format = u->ss.format == PA_SAMPLE_FLOAT32LE || u->ss.format == PA_SAMPLE_FLOAT32BE ? PA_SAMPLE_FLOAT32LE : PA_SAMPLE_S16LE;
+
+ u->performance_mode = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
+ u->input_preset = DEFAULT_INPUT_PRESET;
+ u->set_default = true;
+
+ if (!pa_modargs_get_value_s32(ma, "performance_mode", &performance_mode)) {
+ switch (performance_mode) {
+ case 0:
+ u->performance_mode = AAUDIO_PERFORMANCE_MODE_NONE;
+ break;
+ case 1:
+ u->performance_mode = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
+ break;
+ case 2:
+ u->performance_mode = AAUDIO_PERFORMANCE_MODE_POWER_SAVING;
+ break;
+ }
+ }
+
+ if (pa_modargs_get_value_s32(ma, "input_preset", &u->input_preset) < 0) {
+ LOGW("Failed to parse input_preset argument.");
+ goto error;
+ }
+
+ if (pa_modargs_get_value_boolean(ma, "set_default", &u->set_default) < 0) {
+ LOGW("Failed to parse set_default argument.");
+ goto error;
+ }
+
+ if (pa_create_aaudio_stream(u) < 0) goto error;
+
+ /* The stream is opened here only to discover the device's real format; nothing
+ * is captured until a client connects and the source goes RUNNING. */
+ pa_channel_map_init_extend(&map, u->ss.channels, PA_CHANNEL_MAP_DEFAULT);
+
+ pa_source_new_data_init(&data);
+ data.driver = __FILE__;
+ data.module = m;
+ pa_source_new_data_set_name(&data, pa_modargs_get_value(ma, "source_name", DEFAULT_SOURCE_NAME));
+ pa_source_new_data_set_sample_spec(&data, &u->ss);
+ pa_source_new_data_set_alternate_sample_rate(&data, u->ss.rate);
+ pa_source_new_data_set_channel_map(&data, &map);
+
+ pa_proplist_sets(data.proplist, PA_PROP_DEVICE_DESCRIPTION, _("AAudio Input"));
+ pa_proplist_sets(data.proplist, PA_PROP_DEVICE_CLASS, "abstract");
+
+ if (pa_modargs_get_proplist(ma, "source_properties", data.proplist, PA_UPDATE_REPLACE) < 0) {
+ LOGW("pa_modargs_get_proplist() failed.");
+ pa_source_new_data_done(&data);
+ goto error;
+ }
+
+ u->source = pa_source_new(m->core, &data, PA_SOURCE_HARDWARE);
+ pa_source_new_data_done(&data);
+
+ if (!u->source) {
+ LOGW("Failed to create source object.");
+ goto error;
+ }
+
+ u->source->parent.process_msg = source_process_msg;
+ u->source->set_state_in_io_thread = source_set_state_io_thread;
+ u->source->userdata = u;
+
+ pa_source_set_asyncmsgq(u->source, u->thread_mq.inq);
+ pa_source_set_rtpoll(u->source, u->rtpoll);
+ update_pa_latency(u);
+
+ if (!(u->thread = pa_thread_new("aaudio-source", thread_func, u))) {
+ LOGW("Failed to create thread.");
+ goto error;
+ }
+
+ pa_source_put(u->source);
+
+ /* Outrank AAudioSink.monitor, which PulseAudio creates for every sink and
+ * which Wine would otherwise hand to games as their microphone. */
+ if (u->set_default) {
+ pa_core_set_configured_default_source(m->core, u->source->name);
+ }
+
+ pa_modargs_free(ma);
+ return 0;
+
+error:
+ if (ma) pa_modargs_free(ma);
+ pa__done(m);
+ return -1;
+}