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-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire.inc1
-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire/0010-gst-Implement-new-pwaudio-src-sink-elements-based-on.patch1249
-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire/0011-gst-pwaudioringbuffer-make-the-buffer-size-sensitive.patch60
-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire/0012-gst-pwaudioringbuffer-request-pause-play-on-the-appr.patch76
-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire/0013-gst-pwaudioringbuffer-wait-only-for-STREAM_STATE_CON.patch35
-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire/0014-gst-pwaudiosink-set-the-default-latency-time-buffer-.patch37
-rw-r--r--meta-pipewire/recipes-multimedia/pipewire/pipewire_git.bb5
7 files changed, 1463 insertions, 0 deletions
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire.inc b/meta-pipewire/recipes-multimedia/pipewire/pipewire.inc
index 4a14b07c..e9046e8e 100644
--- a/meta-pipewire/recipes-multimedia/pipewire/pipewire.inc
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire.inc
@@ -21,6 +21,7 @@ PACKAGECONFIG ??= "\
${@bb.utils.filter('DISTRO_FEATURES', 'bluez5', d)} \
alsa audioconvert \
pipewire-alsa \
+ gstreamer \
"
GST_VER = "1.0"
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire/0010-gst-Implement-new-pwaudio-src-sink-elements-based-on.patch b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0010-gst-Implement-new-pwaudio-src-sink-elements-based-on.patch
new file mode 100644
index 00000000..6b1a6441
--- /dev/null
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0010-gst-Implement-new-pwaudio-src-sink-elements-based-on.patch
@@ -0,0 +1,1249 @@
+From bbc875ec4268a88bf2465244e089b119011e7479 Mon Sep 17 00:00:00 2001
+From: George Kiagiadakis <george.kiagiadakis@collabora.com>
+Date: Tue, 19 Feb 2019 18:23:19 +0200
+Subject: [PATCH] gst: Implement new pwaudio{src,sink} elements, based on
+ GstAudioBase{Src,Sink}
+
+These are much more reliable elements to use for audio data.
+* GstAudioBaseSink provides a reliable clock implementation based
+ on the number of samples read/written
+* on the pipewire side we make sure to dequeue, fill and enqueue
+ a single buffer inside the process() function, which avoids
+ underruns
+
+Both elements share a common ringbuffer that actually implements
+the pipewire integration.
+
+Upstream-Status: Submitted [https://github.com/PipeWire/pipewire/pull/140]
+---
+ src/gst/gstpipewire.c | 8 +-
+ src/gst/gstpwaudioringbuffer.c | 542 +++++++++++++++++++++++++++++++++
+ src/gst/gstpwaudioringbuffer.h | 83 +++++
+ src/gst/gstpwaudiosink.c | 200 ++++++++++++
+ src/gst/gstpwaudiosink.h | 48 +++
+ src/gst/gstpwaudiosrc.c | 200 ++++++++++++
+ src/gst/gstpwaudiosrc.h | 48 +++
+ src/gst/meson.build | 6 +
+ 8 files changed, 1134 insertions(+), 1 deletion(-)
+ create mode 100644 src/gst/gstpwaudioringbuffer.c
+ create mode 100644 src/gst/gstpwaudioringbuffer.h
+ create mode 100644 src/gst/gstpwaudiosink.c
+ create mode 100644 src/gst/gstpwaudiosink.h
+ create mode 100644 src/gst/gstpwaudiosrc.c
+ create mode 100644 src/gst/gstpwaudiosrc.h
+
+diff --git a/src/gst/gstpipewire.c b/src/gst/gstpipewire.c
+index 4040264b..68fd446f 100644
+--- a/src/gst/gstpipewire.c
++++ b/src/gst/gstpipewire.c
+@@ -40,6 +40,8 @@
+ #include "gstpipewiresrc.h"
+ #include "gstpipewiresink.h"
+ #include "gstpipewiredeviceprovider.h"
++#include "gstpwaudiosrc.h"
++#include "gstpwaudiosink.h"
+
+ GST_DEBUG_CATEGORY (pipewire_debug);
+
+@@ -52,12 +54,16 @@ plugin_init (GstPlugin *plugin)
+ GST_TYPE_PIPEWIRE_SRC);
+ gst_element_register (plugin, "pipewiresink", GST_RANK_NONE,
+ GST_TYPE_PIPEWIRE_SINK);
++ gst_element_register (plugin, "pwaudiosrc", GST_RANK_NONE,
++ GST_TYPE_PW_AUDIO_SRC);
++ gst_element_register (plugin, "pwaudiosink", GST_RANK_NONE,
++ GST_TYPE_PW_AUDIO_SINK);
+
+ if (!gst_device_provider_register (plugin, "pipewiredeviceprovider",
+ GST_RANK_PRIMARY + 1, GST_TYPE_PIPEWIRE_DEVICE_PROVIDER))
+ return FALSE;
+
+- GST_DEBUG_CATEGORY_INIT (pipewire_debug, "pipewire", 0, "PipeWirie elements");
++ GST_DEBUG_CATEGORY_INIT (pipewire_debug, "pipewire", 0, "PipeWire elements");
+
+ return TRUE;
+ }
+diff --git a/src/gst/gstpwaudioringbuffer.c b/src/gst/gstpwaudioringbuffer.c
+new file mode 100644
+index 00000000..989b2cd7
+--- /dev/null
++++ b/src/gst/gstpwaudioringbuffer.c
+@@ -0,0 +1,542 @@
++/* PipeWire
++ *
++ * Copyright © 2018 Wim Taymans
++ * Copyright © 2019 Collabora Ltd.
++ * @author George Kiagiadakis <george.kiagiadakis@collabora.com>
++ *
++ * Permission is hereby granted, free of charge, to any person obtaining a
++ * copy of this software and associated documentation files (the "Software"),
++ * to deal in the Software without restriction, including without limitation
++ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
++ * and/or sell copies of the Software, and to permit persons to whom the
++ * Software is furnished to do so, subject to the following conditions:
++ *
++ * The above copyright notice and this permission notice (including the next
++ * paragraph) shall be included in all copies or substantial portions of the
++ * Software.
++ *
++ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
++ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
++ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
++ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
++ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
++ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
++ * DEALINGS IN THE SOFTWARE.
++ */
++
++#ifdef HAVE_CONFIG_H
++#include "config.h"
++#endif
++
++#include "gstpwaudioringbuffer.h"
++
++#include <spa/param/audio/format-utils.h>
++#include <spa/pod/builder.h>
++
++GST_DEBUG_CATEGORY_STATIC (pw_audio_ring_buffer_debug);
++#define GST_CAT_DEFAULT pw_audio_ring_buffer_debug
++
++#define gst_pw_audio_ring_buffer_parent_class parent_class
++G_DEFINE_TYPE (GstPwAudioRingBuffer, gst_pw_audio_ring_buffer, GST_TYPE_AUDIO_RING_BUFFER);
++
++enum
++{
++ PROP_0,
++ PROP_ELEMENT,
++ PROP_DIRECTION,
++ PROP_PROPS
++};
++
++static void
++gst_pw_audio_ring_buffer_init (GstPwAudioRingBuffer * self)
++{
++ self->loop = pw_loop_new (NULL);
++ self->main_loop = pw_thread_loop_new (self->loop, "pw-audioringbuffer-loop");
++ self->core = pw_core_new (self->loop, NULL, 0);
++}
++
++static void
++gst_pw_audio_ring_buffer_finalize (GObject * object)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (object);
++
++ pw_core_destroy (self->core);
++ pw_thread_loop_destroy (self->main_loop);
++ pw_loop_destroy (self->loop);
++}
++
++static void
++gst_pw_audio_ring_buffer_set_property (GObject * object, guint prop_id,
++ const GValue * value, GParamSpec * pspec)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (object);
++
++ switch (prop_id) {
++ case PROP_ELEMENT:
++ self->elem = g_value_get_object (value);
++ break;
++
++ case PROP_DIRECTION:
++ self->direction = g_value_get_int (value);
++ break;
++
++ case PROP_PROPS:
++ self->props = g_value_get_pointer (value);
++ break;
++
++ default:
++ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
++ break;
++ }
++}
++
++static void
++on_remote_state_changed (void *data, enum pw_remote_state old,
++ enum pw_remote_state state, const char *error)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (data);
++
++ GST_DEBUG_OBJECT (self->elem, "got remote state %d", state);
++
++ switch (state) {
++ case PW_REMOTE_STATE_UNCONNECTED:
++ case PW_REMOTE_STATE_CONNECTING:
++ case PW_REMOTE_STATE_CONNECTED:
++ break;
++ case PW_REMOTE_STATE_ERROR:
++ GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
++ ("remote error: %s", error), (NULL));
++ break;
++ }
++ pw_thread_loop_signal (self->main_loop, FALSE);
++}
++
++static const struct pw_remote_events remote_events = {
++ PW_VERSION_REMOTE_EVENTS,
++ .state_changed = on_remote_state_changed,
++};
++
++static gboolean
++wait_for_remote_state (GstPwAudioRingBuffer *self,
++ enum pw_remote_state target)
++{
++ while (TRUE) {
++ enum pw_remote_state state = pw_remote_get_state (self->remote, NULL);
++ if (state == target)
++ return TRUE;
++ if (state == PW_REMOTE_STATE_ERROR)
++ return FALSE;
++ pw_thread_loop_wait (self->main_loop);
++ }
++}
++
++static gboolean
++gst_pw_audio_ring_buffer_open_device (GstAudioRingBuffer *buf)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (buf);
++
++ GST_DEBUG_OBJECT (self->elem, "open device");
++
++ if (pw_thread_loop_start (self->main_loop) < 0)
++ goto mainloop_error;
++
++ pw_thread_loop_lock (self->main_loop);
++
++ self->remote = pw_remote_new (self->core, NULL, 0);
++ pw_remote_add_listener (self->remote, &self->remote_listener, &remote_events,
++ self);
++
++ if (self->props->fd == -1)
++ pw_remote_connect (self->remote);
++ else
++ pw_remote_connect_fd (self->remote, self->props->fd);
++
++ GST_DEBUG_OBJECT (self->elem, "waiting for connection");
++
++ if (!wait_for_remote_state (self, PW_REMOTE_STATE_CONNECTED))
++ goto connect_error;
++
++ pw_thread_loop_unlock (self->main_loop);
++
++ return TRUE;
++
++ /* ERRORS */
++mainloop_error:
++ {
++ GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
++ ("Failed to start mainloop"), (NULL));
++ return FALSE;
++ }
++connect_error:
++ {
++ pw_thread_loop_unlock (self->main_loop);
++ return FALSE;
++ }
++}
++
++static gboolean
++gst_pw_audio_ring_buffer_close_device (GstAudioRingBuffer *buf)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (buf);
++
++ GST_DEBUG_OBJECT (self->elem, "closing device");
++
++ pw_thread_loop_lock (self->main_loop);
++ if (self->remote) {
++ pw_remote_disconnect (self->remote);
++ wait_for_remote_state (self, PW_REMOTE_STATE_UNCONNECTED);
++ }
++ pw_thread_loop_unlock (self->main_loop);
++
++ pw_thread_loop_stop (self->main_loop);
++
++ if (self->remote) {
++ pw_remote_destroy (self->remote);
++ self->remote = NULL;
++ }
++ return TRUE;
++}
++
++static void
++on_stream_state_changed (void *data, enum pw_stream_state old,
++ enum pw_stream_state state, const char *error)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (data);
++
++ GST_DEBUG_OBJECT (self->elem, "got stream state: %s",
++ pw_stream_state_as_string (state));
++
++ switch (state) {
++ case PW_STREAM_STATE_UNCONNECTED:
++ GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
++ ("stream disconnected unexpectedly"), (NULL));
++ break;
++ case PW_STREAM_STATE_CONNECTING:
++ case PW_STREAM_STATE_CONFIGURE:
++ case PW_STREAM_STATE_READY:
++ case PW_STREAM_STATE_PAUSED:
++ case PW_STREAM_STATE_STREAMING:
++ break;
++ case PW_STREAM_STATE_ERROR:
++ GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
++ ("stream error: %s", error), (NULL));
++ break;
++ }
++ pw_thread_loop_signal (self->main_loop, FALSE);
++}
++
++static gboolean
++wait_for_stream_state (GstPwAudioRingBuffer *self,
++ enum pw_stream_state target)
++{
++ while (TRUE) {
++ enum pw_stream_state state = pw_stream_get_state (self->stream, NULL);
++ if (state >= target)
++ return TRUE;
++ if (state == PW_STREAM_STATE_ERROR || state == PW_STREAM_STATE_UNCONNECTED)
++ return FALSE;
++ pw_thread_loop_wait (self->main_loop);
++ }
++}
++
++static void
++on_stream_format_changed (void *data, const struct spa_pod *format)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (data);
++ const struct spa_pod *params[1];
++ struct spa_pod_builder b = { NULL };
++ uint8_t buffer[512];
++
++ spa_pod_builder_init (&b, buffer, sizeof (buffer));
++ params[0] = spa_pod_builder_add_object (&b,
++ SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
++ SPA_PARAM_BUFFERS_buffers, SPA_POD_CHOICE_RANGE_Int(16, 1, INT32_MAX),
++ SPA_PARAM_BUFFERS_blocks, SPA_POD_Int(1),
++ SPA_PARAM_BUFFERS_size, SPA_POD_Int(self->segsize),
++ SPA_PARAM_BUFFERS_stride, SPA_POD_Int(self->bpf),
++ SPA_PARAM_BUFFERS_align, SPA_POD_Int(16));
++
++ GST_DEBUG_OBJECT (self->elem, "doing finish format, buffer size:%d", self->segsize);
++ pw_stream_finish_format (self->stream, 0, params, 1);
++}
++
++static void
++on_stream_process (void *data)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (data);
++ GstAudioRingBuffer *buf = GST_AUDIO_RING_BUFFER (data);
++ struct pw_buffer *b;
++ struct spa_data *d;
++ gint size; /*< size to read/write from/to the spa buffer */
++ gint offset; /*< offset to read/write from/to in the spa buffer */
++ gint segment; /*< the current segment number in the ringbuffer */
++ guint8 *ringptr; /*< pointer to the beginning of the current segment */
++ gint segsize; /*< the size of one segment in the ringbuffer */
++ gint copy_size; /*< the bytes to copy in one memcpy() invocation */
++ gint remain; /*< remainder of bytes available in the spa buffer */
++
++ if (g_atomic_int_get (&buf->state) != GST_AUDIO_RING_BUFFER_STATE_STARTED) {
++ GST_LOG_OBJECT (self->elem, "ring buffer is not started");
++ return;
++ }
++
++ b = pw_stream_dequeue_buffer (self->stream);
++ if (!b) {
++ GST_WARNING_OBJECT (self->elem, "no pipewire buffer available");
++ return;
++ }
++
++ d = &b->buffer->datas[0];
++
++ if (self->direction == PW_DIRECTION_OUTPUT) {
++ /* in output mode, always fill the entire spa buffer */
++ offset = d->chunk->offset = 0;
++ size = d->chunk->size = d->maxsize;
++ b->size = size / self->bpf;
++ } else {
++ offset = SPA_MIN (d->chunk->offset, d->maxsize);
++ size = SPA_MIN (d->chunk->size, d->maxsize - offset);
++ }
++
++ do {
++ gst_audio_ring_buffer_prepare_read (buf, &segment, &ringptr, &segsize);
++
++ /* in INPUT (src) mode, it is possible that the skew algorithm
++ * advances the ringbuffer behind our back */
++ if (self->segoffset > 0 && self->cur_segment != segment)
++ self->segoffset = 0;
++
++ copy_size = SPA_MIN (size, segsize - self->segoffset);
++
++ if (self->direction == PW_DIRECTION_OUTPUT) {
++ memcpy (((guint8*) d->data) + offset, ringptr + self->segoffset,
++ copy_size);
++ } else {
++ memcpy (ringptr + self->segoffset, ((guint8*) d->data) + offset,
++ copy_size);
++ }
++
++ remain = size - (segsize - self->segoffset);
++
++ GST_TRACE_OBJECT (self->elem,
++ "seg %d: %s %d bytes remained:%d offset:%d segoffset:%d", segment,
++ self->direction == PW_DIRECTION_INPUT ? "INPUT" : "OUTPUT",
++ copy_size, remain, offset, self->segoffset);
++
++ if (remain >= 0) {
++ offset += (segsize - self->segoffset);
++ size = remain;
++
++ /* write silence on the segment we just read */
++ if (self->direction == PW_DIRECTION_OUTPUT)
++ gst_audio_ring_buffer_clear (buf, segment);
++
++ /* notify that we have read a complete segment */
++ gst_audio_ring_buffer_advance (buf, 1);
++ self->segoffset = 0;
++ } else {
++ self->segoffset += size;
++ self->cur_segment = segment;
++ }
++ } while (remain > 0);
++
++ pw_stream_queue_buffer (self->stream, b);
++}
++
++static const struct pw_stream_events stream_events = {
++ PW_VERSION_STREAM_EVENTS,
++ .state_changed = on_stream_state_changed,
++ .format_changed = on_stream_format_changed,
++ .process = on_stream_process,
++};
++
++static gboolean
++copy_properties (GQuark field_id, const GValue *value, gpointer user_data)
++{
++ struct pw_properties *properties = user_data;
++
++ if (G_VALUE_HOLDS_STRING (value))
++ pw_properties_set (properties,
++ g_quark_to_string (field_id),
++ g_value_get_string (value));
++ return TRUE;
++}
++
++static gboolean
++gst_pw_audio_ring_buffer_acquire (GstAudioRingBuffer *buf,
++ GstAudioRingBufferSpec *spec)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (buf);
++ struct pw_properties *props;
++ struct spa_pod_builder b = { NULL };
++ uint8_t buffer[512];
++ const struct spa_pod *params[1];
++
++ g_return_val_if_fail (spec, FALSE);
++ g_return_val_if_fail (GST_AUDIO_INFO_IS_VALID (&spec->info), FALSE);
++ g_return_val_if_fail (!self->stream, TRUE); /* already acquired */
++
++ g_return_val_if_fail (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW, FALSE);
++ g_return_val_if_fail (GST_AUDIO_INFO_IS_FLOAT (&spec->info), FALSE);
++
++ GST_DEBUG_OBJECT (self->elem, "acquire");
++
++ /* construct param & props objects */
++
++ if (self->props->properties) {
++ props = pw_properties_new (NULL, NULL);
++ gst_structure_foreach (self->props->properties, copy_properties, props);
++ } else {
++ props = NULL;
++ }
++
++ spa_pod_builder_init (&b, buffer, sizeof (buffer));
++ params[0] = spa_pod_builder_add_object (&b,
++ SPA_TYPE_OBJECT_Format, SPA_PARAM_EnumFormat,
++ SPA_FORMAT_mediaType, SPA_POD_Id (SPA_MEDIA_TYPE_audio),
++ SPA_FORMAT_mediaSubtype, SPA_POD_Id (SPA_MEDIA_SUBTYPE_raw),
++ SPA_FORMAT_AUDIO_format, SPA_POD_Id (SPA_AUDIO_FORMAT_F32),
++ SPA_FORMAT_AUDIO_rate, SPA_POD_Int (GST_AUDIO_INFO_RATE (&spec->info)),
++ SPA_FORMAT_AUDIO_channels, SPA_POD_Int (GST_AUDIO_INFO_CHANNELS (&spec->info)));
++
++ self->segsize = spec->segsize;
++ self->bpf = GST_AUDIO_INFO_BPF (&spec->info);
++ self->rate = GST_AUDIO_INFO_RATE (&spec->info);
++ self->segoffset = 0;
++
++ /* connect stream */
++
++ pw_thread_loop_lock (self->main_loop);
++
++ GST_DEBUG_OBJECT (self->elem, "creating stream");
++
++ self->stream = pw_stream_new (self->remote, self->props->client_name, props);
++ pw_stream_add_listener(self->stream, &self->stream_listener, &stream_events,
++ self);
++
++ if (pw_stream_connect (self->stream,
++ self->direction,
++ self->props->path ? (uint32_t)atoi(self->props->path) : SPA_ID_INVALID,
++ PW_STREAM_FLAG_AUTOCONNECT |
++ PW_STREAM_FLAG_MAP_BUFFERS |
++ PW_STREAM_FLAG_RT_PROCESS,
++ params, 1) < 0)
++ goto start_error;
++
++ GST_DEBUG_OBJECT (self->elem, "waiting for stream READY");
++
++ if (!wait_for_stream_state (self, PW_STREAM_STATE_READY))
++ goto start_error;
++
++ pw_thread_loop_unlock (self->main_loop);
++
++ /* allocate the internal ringbuffer */
++
++ spec->seglatency = spec->segtotal + 1;
++ buf->size = spec->segtotal * spec->segsize;
++ buf->memory = g_malloc (buf->size);
++
++ gst_audio_format_fill_silence (buf->spec.info.finfo, buf->memory,
++ buf->size);
++
++ GST_DEBUG_OBJECT (self->elem, "acquire done");
++
++ return TRUE;
++
++start_error:
++ {
++ GST_ERROR_OBJECT (self->elem, "could not start stream");
++ pw_stream_destroy (self->stream);
++ self->stream = NULL;
++ pw_thread_loop_unlock (self->main_loop);
++ return FALSE;
++ }
++}
++
++static gboolean
++gst_pw_audio_ring_buffer_release (GstAudioRingBuffer *buf)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (buf);
++
++ GST_DEBUG_OBJECT (self->elem, "release");
++
++ pw_thread_loop_lock (self->main_loop);
++ if (self->stream) {
++ spa_hook_remove (&self->stream_listener);
++ pw_stream_disconnect (self->stream);
++ pw_stream_destroy (self->stream);
++ self->stream = NULL;
++ }
++ pw_thread_loop_unlock (self->main_loop);
++
++ /* free the buffer */
++ g_free (buf->memory);
++ buf->memory = NULL;
++
++ return TRUE;
++}
++
++static guint
++gst_pw_audio_ring_buffer_delay (GstAudioRingBuffer *buf)
++{
++ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (buf);
++ struct pw_time t;
++
++ if (!self->stream || pw_stream_get_time (self->stream, &t) < 0)
++ return 0;
++
++ if (self->direction == PW_DIRECTION_OUTPUT) {
++ /* on output streams, we set the pw_buffer.size in frames,
++ so no conversion is necessary */
++ return t.queued;
++ } else {
++ /* on input streams, pw_buffer.size is set by pw_stream in ticks,
++ so we need to convert it to frames and also add segoffset, which
++ is the number of bytes we have read but not advertised yet, as
++ the segment is incomplete */
++ if (t.rate.denom > 0)
++ return
++ gst_util_uint64_scale (t.queued, self->rate * t.rate.num, t.rate.denom)
++ + self->segoffset / self->bpf;
++ else
++ return self->segoffset / self->bpf;
++ }
++
++ return 0;
++}
++
++static void
++gst_pw_audio_ring_buffer_class_init (GstPwAudioRingBufferClass * klass)
++{
++ GObjectClass *gobject_class;
++ GstAudioRingBufferClass *gstaudiorbuf_class;
++
++ gobject_class = (GObjectClass *) klass;
++ gstaudiorbuf_class = (GstAudioRingBufferClass *) klass;
++
++ gobject_class->finalize = gst_pw_audio_ring_buffer_finalize;
++ gobject_class->set_property = gst_pw_audio_ring_buffer_set_property;
++
++ gstaudiorbuf_class->open_device = gst_pw_audio_ring_buffer_open_device;
++ gstaudiorbuf_class->acquire = gst_pw_audio_ring_buffer_acquire;
++ gstaudiorbuf_class->release = gst_pw_audio_ring_buffer_release;
++ gstaudiorbuf_class->close_device = gst_pw_audio_ring_buffer_close_device;
++ gstaudiorbuf_class->delay = gst_pw_audio_ring_buffer_delay;
++
++ g_object_class_install_property (gobject_class, PROP_ELEMENT,
++ g_param_spec_object ("element", "Element", "The audio source or sink",
++ GST_TYPE_ELEMENT,
++ G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_DIRECTION,
++ g_param_spec_int ("direction", "Direction", "The stream direction",
++ PW_DIRECTION_INPUT, PW_DIRECTION_OUTPUT, PW_DIRECTION_INPUT,
++ G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_PROPS,
++ g_param_spec_pointer ("props", "Properties", "The properties struct",
++ G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
++
++ GST_DEBUG_CATEGORY_INIT (pw_audio_ring_buffer_debug, "pwaudioringbuffer", 0,
++ "PipeWire Audio Ring Buffer");
++}
+diff --git a/src/gst/gstpwaudioringbuffer.h b/src/gst/gstpwaudioringbuffer.h
+new file mode 100644
+index 00000000..f47f668a
+--- /dev/null
++++ b/src/gst/gstpwaudioringbuffer.h
+@@ -0,0 +1,83 @@
++/* PipeWire
++ *
++ * Copyright © 2018 Wim Taymans
++ * Copyright © 2019 Collabora Ltd.
++ * @author George Kiagiadakis <george.kiagiadakis@collabora.com>
++ *
++ * Permission is hereby granted, free of charge, to any person obtaining a
++ * copy of this software and associated documentation files (the "Software"),
++ * to deal in the Software without restriction, including without limitation
++ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
++ * and/or sell copies of the Software, and to permit persons to whom the
++ * Software is furnished to do so, subject to the following conditions:
++ *
++ * The above copyright notice and this permission notice (including the next
++ * paragraph) shall be included in all copies or substantial portions of the
++ * Software.
++ *
++ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
++ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
++ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
++ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
++ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
++ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
++ * DEALINGS IN THE SOFTWARE.
++ */
++
++#ifndef __GST_PW_AUDIO_RING_BUFFER_H__
++#define __GST_PW_AUDIO_RING_BUFFER_H__
++
++#include <gst/gst.h>
++#include <gst/audio/audio.h>
++#include <pipewire/pipewire.h>
++
++G_BEGIN_DECLS
++
++#define GST_TYPE_PW_AUDIO_RING_BUFFER \
++ (gst_pw_audio_ring_buffer_get_type ())
++
++G_DECLARE_FINAL_TYPE(GstPwAudioRingBuffer, gst_pw_audio_ring_buffer,
++ GST, PW_AUDIO_RING_BUFFER, GstAudioRingBuffer);
++
++typedef struct _GstPwAudioRingBufferProps GstPwAudioRingBufferProps;
++
++struct _GstPwAudioRingBuffer
++{
++ GstAudioRingBuffer parent;
++
++ /* properties */
++ GstElement *elem;
++ enum pw_direction direction;
++ GstPwAudioRingBufferProps *props;
++
++ /* internal */
++ struct pw_loop *loop;
++ struct pw_thread_loop *main_loop;
++
++ struct pw_core *core;
++ struct pw_remote *remote;
++ struct spa_hook remote_listener;
++
++ struct pw_stream *stream;
++ struct spa_hook stream_listener;
++
++ gint segsize;
++ gint bpf;
++ gint rate;
++
++ /* on_stream_process() state */
++ gint segoffset;
++ gint cur_segment;
++};
++
++struct _GstPwAudioRingBufferProps
++{
++ gchar *path;
++ gchar *client_name;
++ GstStructure *properties;
++ int fd;
++};
++
++G_END_DECLS
++
++#endif
+diff --git a/src/gst/gstpwaudiosink.c b/src/gst/gstpwaudiosink.c
+new file mode 100644
+index 00000000..6cb71385
+--- /dev/null
++++ b/src/gst/gstpwaudiosink.c
+@@ -0,0 +1,200 @@
++/* PipeWire
++ *
++ * Copyright © 2018 Wim Taymans
++ * Copyright © 2019 Collabora Ltd.
++ * @author George Kiagiadakis <george.kiagiadakis@collabora.com>
++ *
++ * Permission is hereby granted, free of charge, to any person obtaining a
++ * copy of this software and associated documentation files (the "Software"),
++ * to deal in the Software without restriction, including without limitation
++ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
++ * and/or sell copies of the Software, and to permit persons to whom the
++ * Software is furnished to do so, subject to the following conditions:
++ *
++ * The above copyright notice and this permission notice (including the next
++ * paragraph) shall be included in all copies or substantial portions of the
++ * Software.
++ *
++ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
++ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
++ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
++ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
++ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
++ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
++ * DEALINGS IN THE SOFTWARE.
++ */
++
++#ifdef HAVE_CONFIG_H
++#include "config.h"
++#endif
++
++#include "gstpwaudiosink.h"
++
++GST_DEBUG_CATEGORY_STATIC (pw_audio_sink_debug);
++#define GST_CAT_DEFAULT pw_audio_sink_debug
++
++G_DEFINE_TYPE (GstPwAudioSink, gst_pw_audio_sink, GST_TYPE_AUDIO_BASE_SINK);
++
++enum
++{
++ PROP_0,
++ PROP_PATH,
++ PROP_CLIENT_NAME,
++ PROP_STREAM_PROPERTIES,
++ PROP_FD
++};
++
++static GstStaticPadTemplate gst_pw_audio_sink_template =
++GST_STATIC_PAD_TEMPLATE ("sink",
++ GST_PAD_SINK,
++ GST_PAD_ALWAYS,
++ GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_NE (F32))
++ ", layout = (string)\"interleaved\"")
++);
++
++
++static void
++gst_pw_audio_sink_init (GstPwAudioSink * self)
++{
++ self->props.fd = -1;
++}
++
++static void
++gst_pw_audio_sink_finalize (GObject * object)
++{
++ GstPwAudioSink *pwsink = GST_PW_AUDIO_SINK (object);
++
++ g_free (pwsink->props.path);
++ g_free (pwsink->props.client_name);
++ if (pwsink->props.properties)
++ gst_structure_free (pwsink->props.properties);
++}
++
++static void
++gst_pw_audio_sink_set_property (GObject * object, guint prop_id,
++ const GValue * value, GParamSpec * pspec)
++{
++ GstPwAudioSink *pwsink = GST_PW_AUDIO_SINK (object);
++
++ switch (prop_id) {
++ case PROP_PATH:
++ g_free (pwsink->props.path);
++ pwsink->props.path = g_value_dup_string (value);
++ break;
++
++ case PROP_CLIENT_NAME:
++ g_free (pwsink->props.client_name);
++ pwsink->props.client_name = g_value_dup_string (value);
++ break;
++
++ case PROP_STREAM_PROPERTIES:
++ if (pwsink->props.properties)
++ gst_structure_free (pwsink->props.properties);
++ pwsink->props.properties =
++ gst_structure_copy (gst_value_get_structure (value));
++ break;
++
++ case PROP_FD:
++ pwsink->props.fd = g_value_get_int (value);
++ break;
++
++ default:
++ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
++ break;
++ }
++}
++
++static void
++gst_pw_audio_sink_get_property (GObject * object, guint prop_id,
++ GValue * value, GParamSpec * pspec)
++{
++ GstPwAudioSink *pwsink = GST_PW_AUDIO_SINK (object);
++
++ switch (prop_id) {
++ case PROP_PATH:
++ g_value_set_string (value, pwsink->props.path);
++ break;
++
++ case PROP_CLIENT_NAME:
++ g_value_set_string (value, pwsink->props.client_name);
++ break;
++
++ case PROP_STREAM_PROPERTIES:
++ gst_value_set_structure (value, pwsink->props.properties);
++ break;
++
++ case PROP_FD:
++ g_value_set_int (value, pwsink->props.fd);
++ break;
++
++ default:
++ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
++ break;
++ }
++}
++
++static GstAudioRingBuffer *
++gst_pw_audio_sink_create_ringbuffer (GstAudioBaseSink * sink)
++{
++ GstPwAudioSink *self = GST_PW_AUDIO_SINK (sink);
++ GstAudioRingBuffer *buffer;
++
++ GST_DEBUG_OBJECT (sink, "creating ringbuffer");
++ buffer = g_object_new (GST_TYPE_PW_AUDIO_RING_BUFFER,
++ "element", sink,
++ "direction", PW_DIRECTION_OUTPUT,
++ "props", &self->props,
++ NULL);
++ GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
++
++ return buffer;
++}
++
++static void
++gst_pw_audio_sink_class_init (GstPwAudioSinkClass * klass)
++{
++ GObjectClass *gobject_class;
++ GstElementClass *gstelement_class;
++ GstAudioBaseSinkClass *gstaudiobsink_class;
++
++ gobject_class = (GObjectClass *) klass;
++ gstelement_class = (GstElementClass *) klass;
++ gstaudiobsink_class = (GstAudioBaseSinkClass *) klass;
++
++ gobject_class->finalize = gst_pw_audio_sink_finalize;
++ gobject_class->set_property = gst_pw_audio_sink_set_property;
++ gobject_class->get_property = gst_pw_audio_sink_get_property;
++
++ gstaudiobsink_class->create_ringbuffer = gst_pw_audio_sink_create_ringbuffer;
++
++ g_object_class_install_property (gobject_class, PROP_PATH,
++ g_param_spec_string ("path", "Path",
++ "The sink path to connect to (NULL = default)", NULL,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
++ g_param_spec_string ("client-name", "Client Name",
++ "The client name to use (NULL = default)", NULL,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_STREAM_PROPERTIES,
++ g_param_spec_boxed ("stream-properties", "Stream properties",
++ "List of PipeWire stream properties", GST_TYPE_STRUCTURE,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_FD,
++ g_param_spec_int ("fd", "Fd", "The fd to connect with", -1, G_MAXINT, -1,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ gst_element_class_set_static_metadata (gstelement_class,
++ "PipeWire Audio sink", "Sink/Audio",
++ "Send audio to PipeWire",
++ "George Kiagiadakis <george.kiagiadakis@collabora.com>");
++
++ gst_element_class_add_pad_template (gstelement_class,
++ gst_static_pad_template_get (&gst_pw_audio_sink_template));
++
++ GST_DEBUG_CATEGORY_INIT (pw_audio_sink_debug, "pwaudiosink", 0,
++ "PipeWire Audio Sink");
++}
++
+diff --git a/src/gst/gstpwaudiosink.h b/src/gst/gstpwaudiosink.h
+new file mode 100644
+index 00000000..7ed0de7b
+--- /dev/null
++++ b/src/gst/gstpwaudiosink.h
+@@ -0,0 +1,48 @@
++/* PipeWire
++ *
++ * Copyright © 2018 Wim Taymans
++ * Copyright © 2019 Collabora Ltd.
++ * @author George Kiagiadakis <george.kiagiadakis@collabora.com>
++ *
++ * Permission is hereby granted, free of charge, to any person obtaining a
++ * copy of this software and associated documentation files (the "Software"),
++ * to deal in the Software without restriction, including without limitation
++ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
++ * and/or sell copies of the Software, and to permit persons to whom the
++ * Software is furnished to do so, subject to the following conditions:
++ *
++ * The above copyright notice and this permission notice (including the next
++ * paragraph) shall be included in all copies or substantial portions of the
++ * Software.
++ *
++ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
++ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
++ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
++ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
++ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
++ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
++ * DEALINGS IN THE SOFTWARE.
++ */
++
++#ifndef __GST_PW_AUDIO_SINK_H__
++#define __GST_PW_AUDIO_SINK_H__
++
++#include "gstpwaudioringbuffer.h"
++
++G_BEGIN_DECLS
++
++#define GST_TYPE_PW_AUDIO_SINK \
++ (gst_pw_audio_sink_get_type ())
++
++G_DECLARE_FINAL_TYPE(GstPwAudioSink, gst_pw_audio_sink,
++ GST, PW_AUDIO_SINK, GstAudioBaseSink);
++
++struct _GstPwAudioSink
++{
++ GstAudioBaseSink parent;
++ GstPwAudioRingBufferProps props;
++};
++
++G_END_DECLS
++
++#endif
+diff --git a/src/gst/gstpwaudiosrc.c b/src/gst/gstpwaudiosrc.c
+new file mode 100644
+index 00000000..6c522982
+--- /dev/null
++++ b/src/gst/gstpwaudiosrc.c
+@@ -0,0 +1,200 @@
++/* PipeWire
++ *
++ * Copyright © 2018 Wim Taymans
++ * Copyright © 2019 Collabora Ltd.
++ * @author George Kiagiadakis <george.kiagiadakis@collabora.com>
++ *
++ * Permission is hereby granted, free of charge, to any person obtaining a
++ * copy of this software and associated documentation files (the "Software"),
++ * to deal in the Software without restriction, including without limitation
++ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
++ * and/or sell copies of the Software, and to permit persons to whom the
++ * Software is furnished to do so, subject to the following conditions:
++ *
++ * The above copyright notice and this permission notice (including the next
++ * paragraph) shall be included in all copies or substantial portions of the
++ * Software.
++ *
++ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
++ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
++ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
++ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
++ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
++ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
++ * DEALINGS IN THE SOFTWARE.
++ */
++
++#ifdef HAVE_CONFIG_H
++#include "config.h"
++#endif
++
++#include "gstpwaudiosrc.h"
++
++GST_DEBUG_CATEGORY_STATIC (pw_audio_src_debug);
++#define GST_CAT_DEFAULT pw_audio_src_debug
++
++G_DEFINE_TYPE (GstPwAudioSrc, gst_pw_audio_src, GST_TYPE_AUDIO_BASE_SRC);
++
++enum
++{
++ PROP_0,
++ PROP_PATH,
++ PROP_CLIENT_NAME,
++ PROP_STREAM_PROPERTIES,
++ PROP_FD
++};
++
++static GstStaticPadTemplate gst_pw_audio_src_template =
++GST_STATIC_PAD_TEMPLATE ("src",
++ GST_PAD_SRC,
++ GST_PAD_ALWAYS,
++ GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_NE (F32))
++ ", layout = (string)\"interleaved\"")
++);
++
++
++static void
++gst_pw_audio_src_init (GstPwAudioSrc * self)
++{
++ self->props.fd = -1;
++}
++
++static void
++gst_pw_audio_src_finalize (GObject * object)
++{
++ GstPwAudioSrc *self = GST_PW_AUDIO_SRC (object);
++
++ g_free (self->props.path);
++ g_free (self->props.client_name);
++ if (self->props.properties)
++ gst_structure_free (self->props.properties);
++}
++
++static void
++gst_pw_audio_src_set_property (GObject * object, guint prop_id,
++ const GValue * value, GParamSpec * pspec)
++{
++ GstPwAudioSrc *self = GST_PW_AUDIO_SRC (object);
++
++ switch (prop_id) {
++ case PROP_PATH:
++ g_free (self->props.path);
++ self->props.path = g_value_dup_string (value);
++ break;
++
++ case PROP_CLIENT_NAME:
++ g_free (self->props.client_name);
++ self->props.client_name = g_value_dup_string (value);
++ break;
++
++ case PROP_STREAM_PROPERTIES:
++ if (self->props.properties)
++ gst_structure_free (self->props.properties);
++ self->props.properties =
++ gst_structure_copy (gst_value_get_structure (value));
++ break;
++
++ case PROP_FD:
++ self->props.fd = g_value_get_int (value);
++ break;
++
++ default:
++ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
++ break;
++ }
++}
++
++static void
++gst_pw_audio_src_get_property (GObject * object, guint prop_id,
++ GValue * value, GParamSpec * pspec)
++{
++ GstPwAudioSrc *self = GST_PW_AUDIO_SRC (object);
++
++ switch (prop_id) {
++ case PROP_PATH:
++ g_value_set_string (value, self->props.path);
++ break;
++
++ case PROP_CLIENT_NAME:
++ g_value_set_string (value, self->props.client_name);
++ break;
++
++ case PROP_STREAM_PROPERTIES:
++ gst_value_set_structure (value, self->props.properties);
++ break;
++
++ case PROP_FD:
++ g_value_set_int (value, self->props.fd);
++ break;
++
++ default:
++ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
++ break;
++ }
++}
++
++static GstAudioRingBuffer *
++gst_pw_audio_src_create_ringbuffer (GstAudioBaseSrc * sink)
++{
++ GstPwAudioSrc *self = GST_PW_AUDIO_SRC (sink);
++ GstAudioRingBuffer *buffer;
++
++ GST_DEBUG_OBJECT (sink, "creating ringbuffer");
++ buffer = g_object_new (GST_TYPE_PW_AUDIO_RING_BUFFER,
++ "element", sink,
++ "direction", PW_DIRECTION_INPUT,
++ "props", &self->props,
++ NULL);
++ GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
++
++ return buffer;
++}
++
++static void
++gst_pw_audio_src_class_init (GstPwAudioSrcClass * klass)
++{
++ GObjectClass *gobject_class;
++ GstElementClass *gstelement_class;
++ GstAudioBaseSrcClass *gstaudiobsrc_class;
++
++ gobject_class = (GObjectClass *) klass;
++ gstelement_class = (GstElementClass *) klass;
++ gstaudiobsrc_class = (GstAudioBaseSrcClass *) klass;
++
++ gobject_class->finalize = gst_pw_audio_src_finalize;
++ gobject_class->set_property = gst_pw_audio_src_set_property;
++ gobject_class->get_property = gst_pw_audio_src_get_property;
++
++ gstaudiobsrc_class->create_ringbuffer = gst_pw_audio_src_create_ringbuffer;
++
++ g_object_class_install_property (gobject_class, PROP_PATH,
++ g_param_spec_string ("path", "Path",
++ "The sink path to connect to (NULL = default)", NULL,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
++ g_param_spec_string ("client-name", "Client Name",
++ "The client name to use (NULL = default)", NULL,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_STREAM_PROPERTIES,
++ g_param_spec_boxed ("stream-properties", "Stream properties",
++ "List of PipeWire stream properties", GST_TYPE_STRUCTURE,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ g_object_class_install_property (gobject_class, PROP_FD,
++ g_param_spec_int ("fd", "Fd", "The fd to connect with", -1, G_MAXINT, -1,
++ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ gst_element_class_set_static_metadata (gstelement_class,
++ "PipeWire Audio source", "Source/Audio",
++ "Receive audio from PipeWire",
++ "George Kiagiadakis <george.kiagiadakis@collabora.com>");
++
++ gst_element_class_add_pad_template (gstelement_class,
++ gst_static_pad_template_get (&gst_pw_audio_src_template));
++
++ GST_DEBUG_CATEGORY_INIT (pw_audio_src_debug, "pwaudiosrc", 0,
++ "PipeWire Audio Src");
++}
++
+diff --git a/src/gst/gstpwaudiosrc.h b/src/gst/gstpwaudiosrc.h
+new file mode 100644
+index 00000000..c46e644c
+--- /dev/null
++++ b/src/gst/gstpwaudiosrc.h
+@@ -0,0 +1,48 @@
++/* PipeWire
++ *
++ * Copyright © 2018 Wim Taymans
++ * Copyright © 2019 Collabora Ltd.
++ * @author George Kiagiadakis <george.kiagiadakis@collabora.com>
++ *
++ * Permission is hereby granted, free of charge, to any person obtaining a
++ * copy of this software and associated documentation files (the "Software"),
++ * to deal in the Software without restriction, including without limitation
++ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
++ * and/or sell copies of the Software, and to permit persons to whom the
++ * Software is furnished to do so, subject to the following conditions:
++ *
++ * The above copyright notice and this permission notice (including the next
++ * paragraph) shall be included in all copies or substantial portions of the
++ * Software.
++ *
++ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
++ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
++ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
++ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
++ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
++ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
++ * DEALINGS IN THE SOFTWARE.
++ */
++
++#ifndef __GST_PW_AUDIO_SRC_H__
++#define __GST_PW_AUDIO_SRC_H__
++
++#include "gstpwaudioringbuffer.h"
++
++G_BEGIN_DECLS
++
++#define GST_TYPE_PW_AUDIO_SRC \
++ (gst_pw_audio_src_get_type ())
++
++G_DECLARE_FINAL_TYPE(GstPwAudioSrc, gst_pw_audio_src,
++ GST, PW_AUDIO_SRC, GstAudioBaseSrc);
++
++struct _GstPwAudioSrc
++{
++ GstAudioBaseSrc parent;
++ GstPwAudioRingBufferProps props;
++};
++
++G_END_DECLS
++
++#endif
+diff --git a/src/gst/meson.build b/src/gst/meson.build
+index ad0e0801..0e922347 100644
+--- a/src/gst/meson.build
++++ b/src/gst/meson.build
+@@ -6,6 +6,9 @@ pipewire_gst_sources = [
+ 'gstpipewirepool.c',
+ 'gstpipewiresink.c',
+ 'gstpipewiresrc.c',
++ 'gstpwaudioringbuffer.c',
++ 'gstpwaudiosink.c',
++ 'gstpwaudiosrc.c',
+ ]
+
+ pipewire_gst_headers = [
+@@ -15,6 +18,9 @@ pipewire_gst_headers = [
+ 'gstpipewirepool.h',
+ 'gstpipewiresink.h',
+ 'gstpipewiresrc.h',
++ 'gstpwaudioringbuffer.h',
++ 'gstpwaudiosink.h',
++ 'gstpwaudiosrc.h',
+ ]
+
+ pipewire_gst_c_args = [
+--
+2.20.1
+
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire/0011-gst-pwaudioringbuffer-make-the-buffer-size-sensitive.patch b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0011-gst-pwaudioringbuffer-make-the-buffer-size-sensitive.patch
new file mode 100644
index 00000000..5ffabb6d
--- /dev/null
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0011-gst-pwaudioringbuffer-make-the-buffer-size-sensitive.patch
@@ -0,0 +1,60 @@
+From 6e289d0058d71bc433d1918a8bbf3305f3e4f517 Mon Sep 17 00:00:00 2001
+From: Julian Bouzas <julian.bouzas@collabora.com>
+Date: Tue, 7 May 2019 10:36:35 -0400
+Subject: [PATCH] gst/pwaudioringbuffer: make the buffer size sensitive to the
+ number of channels
+
+Upstream-Status: Submitted [https://github.com/PipeWire/pipewire/pull/140]
+---
+ src/gst/gstpwaudioringbuffer.c | 6 ++++--
+ src/gst/gstpwaudioringbuffer.h | 1 +
+ 2 files changed, 5 insertions(+), 2 deletions(-)
+
+diff --git a/src/gst/gstpwaudioringbuffer.c b/src/gst/gstpwaudioringbuffer.c
+index 989b2cd7..181304e8 100644
+--- a/src/gst/gstpwaudioringbuffer.c
++++ b/src/gst/gstpwaudioringbuffer.c
+@@ -246,17 +246,18 @@ on_stream_format_changed (void *data, const struct spa_pod *format)
+ const struct spa_pod *params[1];
+ struct spa_pod_builder b = { NULL };
+ uint8_t buffer[512];
++ const gint b_size = self->segsize * self->channels;
+
+ spa_pod_builder_init (&b, buffer, sizeof (buffer));
+ params[0] = spa_pod_builder_add_object (&b,
+ SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
+ SPA_PARAM_BUFFERS_buffers, SPA_POD_CHOICE_RANGE_Int(16, 1, INT32_MAX),
+ SPA_PARAM_BUFFERS_blocks, SPA_POD_Int(1),
+- SPA_PARAM_BUFFERS_size, SPA_POD_Int(self->segsize),
++ SPA_PARAM_BUFFERS_size, SPA_POD_Int(b_size),
+ SPA_PARAM_BUFFERS_stride, SPA_POD_Int(self->bpf),
+ SPA_PARAM_BUFFERS_align, SPA_POD_Int(16));
+
+- GST_DEBUG_OBJECT (self->elem, "doing finish format, buffer size:%d", self->segsize);
++ GST_DEBUG_OBJECT (self->elem, "doing finish format, buffer size:%d", b_size);
+ pw_stream_finish_format (self->stream, 0, params, 1);
+ }
+
+@@ -402,6 +403,7 @@ gst_pw_audio_ring_buffer_acquire (GstAudioRingBuffer *buf,
+ self->segsize = spec->segsize;
+ self->bpf = GST_AUDIO_INFO_BPF (&spec->info);
+ self->rate = GST_AUDIO_INFO_RATE (&spec->info);
++ self->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
+ self->segoffset = 0;
+
+ /* connect stream */
+diff --git a/src/gst/gstpwaudioringbuffer.h b/src/gst/gstpwaudioringbuffer.h
+index f47f668a..f600f012 100644
+--- a/src/gst/gstpwaudioringbuffer.h
++++ b/src/gst/gstpwaudioringbuffer.h
+@@ -64,6 +64,7 @@ struct _GstPwAudioRingBuffer
+ gint segsize;
+ gint bpf;
+ gint rate;
++ gint channels;
+
+ /* on_stream_process() state */
+ gint segoffset;
+--
+2.20.1
+
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire/0012-gst-pwaudioringbuffer-request-pause-play-on-the-appr.patch b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0012-gst-pwaudioringbuffer-request-pause-play-on-the-appr.patch
new file mode 100644
index 00000000..3680cc35
--- /dev/null
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0012-gst-pwaudioringbuffer-request-pause-play-on-the-appr.patch
@@ -0,0 +1,76 @@
+From 1eb1e3a839f97ad4aa43c289f702c587a068a333 Mon Sep 17 00:00:00 2001
+From: George Kiagiadakis <george.kiagiadakis@collabora.com>
+Date: Thu, 11 Jul 2019 16:21:17 +0300
+Subject: [PATCH] gst/pwaudioringbuffer: request pause/play on the appropriate
+ stream state changes
+
+This allows the client to properly go to PAUSED when the session manager
+unlinks the stream and go again to PLAYING when the sm re-links it.
+This allows the session manager to implement policies without letting
+the client pipeline freeze (in the absence of a running audio clock)
+when it is unlinked. Note that in case the client doesn't handle the
+request, there is still no issue. Like in pulseaudio, the clock just
+freezes, so the pipeline stops progressing.
+
+This is similar to the pulseaudio cork/uncork mechanism.
+
+Upstream-Status: Submitted [https://github.com/PipeWire/pipewire/pull/140]
+---
+ src/gst/gstpwaudioringbuffer.c | 27 +++++++++++++++++++++++----
+ 1 file changed, 23 insertions(+), 4 deletions(-)
+
+diff --git a/src/gst/gstpwaudioringbuffer.c b/src/gst/gstpwaudioringbuffer.c
+index 181304e8..04259927 100644
+--- a/src/gst/gstpwaudioringbuffer.c
++++ b/src/gst/gstpwaudioringbuffer.c
+@@ -202,11 +202,16 @@ on_stream_state_changed (void *data, enum pw_stream_state old,
+ enum pw_stream_state state, const char *error)
+ {
+ GstPwAudioRingBuffer *self = GST_PW_AUDIO_RING_BUFFER (data);
++ GstMessage *msg;
+
+ GST_DEBUG_OBJECT (self->elem, "got stream state: %s",
+ pw_stream_state_as_string (state));
+
+ switch (state) {
++ case PW_STREAM_STATE_ERROR:
++ GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
++ ("stream error: %s", error), (NULL));
++ break;
+ case PW_STREAM_STATE_UNCONNECTED:
+ GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
+ ("stream disconnected unexpectedly"), (NULL));
+@@ -214,12 +219,26 @@ on_stream_state_changed (void *data, enum pw_stream_state old,
+ case PW_STREAM_STATE_CONNECTING:
+ case PW_STREAM_STATE_CONFIGURE:
+ case PW_STREAM_STATE_READY:
++ break;
+ case PW_STREAM_STATE_PAUSED:
+- case PW_STREAM_STATE_STREAMING:
++ if (old == PW_STREAM_STATE_STREAMING) {
++ if (GST_STATE (self->elem) != GST_STATE_PAUSED &&
++ GST_STATE_TARGET (self->elem) != GST_STATE_PAUSED) {
++ GST_DEBUG_OBJECT (self->elem, "requesting GST_STATE_PAUSED");
++ msg = gst_message_new_request_state (GST_OBJECT (self->elem),
++ GST_STATE_PAUSED);
++ gst_element_post_message (self->elem, msg);
++ }
++ }
+ break;
+- case PW_STREAM_STATE_ERROR:
+- GST_ELEMENT_ERROR (self->elem, RESOURCE, FAILED,
+- ("stream error: %s", error), (NULL));
++ case PW_STREAM_STATE_STREAMING:
++ if (GST_STATE (self->elem) != GST_STATE_PLAYING &&
++ GST_STATE_TARGET (self->elem) != GST_STATE_PLAYING) {
++ GST_DEBUG_OBJECT (self->elem, "requesting GST_STATE_PLAYING");
++ msg = gst_message_new_request_state (GST_OBJECT (self->elem),
++ GST_STATE_PLAYING);
++ gst_element_post_message (self->elem, msg);
++ }
+ break;
+ }
+ pw_thread_loop_signal (self->main_loop, FALSE);
+--
+2.20.1
+
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire/0013-gst-pwaudioringbuffer-wait-only-for-STREAM_STATE_CON.patch b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0013-gst-pwaudioringbuffer-wait-only-for-STREAM_STATE_CON.patch
new file mode 100644
index 00000000..539e3a5e
--- /dev/null
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0013-gst-pwaudioringbuffer-wait-only-for-STREAM_STATE_CON.patch
@@ -0,0 +1,35 @@
+From 1b2bf0f435f2912c32fbd7a6118ed9bfb41f031c Mon Sep 17 00:00:00 2001
+From: George Kiagiadakis <george.kiagiadakis@collabora.com>
+Date: Thu, 11 Jul 2019 16:34:35 +0300
+Subject: [PATCH] gst/pwaudioringbuffer: wait only for STREAM_STATE_CONFIGURE
+ when starting
+
+The CONFIGURE state is reached when the pw_client_node is exported,
+while the READY state requires the session manager to try and link
+the stream. If the SM does not want to link the stream due to policy,
+the client should not hang there forever.
+
+Upstream-Status: Submitted [https://github.com/PipeWire/pipewire/pull/140]
+---
+ src/gst/gstpwaudioringbuffer.c | 4 ++--
+ 1 file changed, 2 insertions(+), 2 deletions(-)
+
+diff --git a/src/gst/gstpwaudioringbuffer.c b/src/gst/gstpwaudioringbuffer.c
+index 04259927..b92b5feb 100644
+--- a/src/gst/gstpwaudioringbuffer.c
++++ b/src/gst/gstpwaudioringbuffer.c
+@@ -444,9 +444,9 @@ gst_pw_audio_ring_buffer_acquire (GstAudioRingBuffer *buf,
+ params, 1) < 0)
+ goto start_error;
+
+- GST_DEBUG_OBJECT (self->elem, "waiting for stream READY");
++ GST_DEBUG_OBJECT (self->elem, "waiting for stream CONFIGURE");
+
+- if (!wait_for_stream_state (self, PW_STREAM_STATE_READY))
++ if (!wait_for_stream_state (self, PW_STREAM_STATE_CONFIGURE))
+ goto start_error;
+
+ pw_thread_loop_unlock (self->main_loop);
+--
+2.20.1
+
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire/0014-gst-pwaudiosink-set-the-default-latency-time-buffer-.patch b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0014-gst-pwaudiosink-set-the-default-latency-time-buffer-.patch
new file mode 100644
index 00000000..6f15b7f7
--- /dev/null
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire/0014-gst-pwaudiosink-set-the-default-latency-time-buffer-.patch
@@ -0,0 +1,37 @@
+From 460ce06c9cc6fd7b0106e0ce8a265bbeff4ae406 Mon Sep 17 00:00:00 2001
+From: George Kiagiadakis <george.kiagiadakis@collabora.com>
+Date: Thu, 11 Jul 2019 17:07:15 +0300
+Subject: [PATCH] gst/pwaudiosink: set the default latency time (buffer size)
+ to be 21.3ms
+
+This is to solve underrun issues that seem to appear with the default
+10ms latency that GstBaseAudioSink has.
+Hopefully in the future we will have a better mechanism to pick
+the appropriate latency instead of hardcoding it here.
+
+Upstream-Status: Submitted [https://github.com/PipeWire/pipewire/pull/140]
+---
+ src/gst/gstpwaudiosink.c | 7 +++++++
+ 1 file changed, 7 insertions(+)
+
+diff --git a/src/gst/gstpwaudiosink.c b/src/gst/gstpwaudiosink.c
+index 6cb71385..069996c3 100644
+--- a/src/gst/gstpwaudiosink.c
++++ b/src/gst/gstpwaudiosink.c
+@@ -57,6 +57,13 @@ static void
+ gst_pw_audio_sink_init (GstPwAudioSink * self)
+ {
+ self->props.fd = -1;
++
++ /* Bump the default buffer size up to 21.3 ms, which is the default on most
++ * sound cards, in hope to match the alsa buffer size on the pipewire server.
++ * This may not always happen, but it still sounds better than the 10ms
++ * default latency. This is temporary until we have a better mechanism to
++ * select the appropriate latency */
++ GST_AUDIO_BASE_SINK (self)->latency_time = 21333;
+ }
+
+ static void
+--
+2.20.1
+
diff --git a/meta-pipewire/recipes-multimedia/pipewire/pipewire_git.bb b/meta-pipewire/recipes-multimedia/pipewire/pipewire_git.bb
index dd1eebcc..43aae8ea 100644
--- a/meta-pipewire/recipes-multimedia/pipewire/pipewire_git.bb
+++ b/meta-pipewire/recipes-multimedia/pipewire/pipewire_git.bb
@@ -10,6 +10,11 @@ SRC_URI = "gitsm://github.com/PipeWire/pipewire;protocol=https;branch=work \
file://0007-alsa-make-corrections-on-the-timeout-based-on-how-fa.patch \
file://0008-audio-dsp-allow-mode-to-be-set-with-a-property.patch \
file://0009-audioconvert-do-setup-internal-links-and-buffers-als.patch \
+ file://0010-gst-Implement-new-pwaudio-src-sink-elements-based-on.patch \
+ file://0011-gst-pwaudioringbuffer-make-the-buffer-size-sensitive.patch \
+ file://0012-gst-pwaudioringbuffer-request-pause-play-on-the-appr.patch \
+ file://0013-gst-pwaudioringbuffer-wait-only-for-STREAM_STATE_CON.patch \
+ file://0014-gst-pwaudiosink-set-the-default-latency-time-buffer-.patch \
"
SRCREV = "4be788962e60891237f1f018627bf709ae3981e6"