diff options
Diffstat (limited to 'meta-pipewire')
8 files changed, 1464 insertions, 0 deletions
diff --git a/meta-pipewire/recipes-core/packagegroups/packagegroup-pipewire.bb b/meta-pipewire/recipes-core/packagegroups/packagegroup-pipewire.bb index 285caca2..470310ad 100644 --- a/meta-pipewire/recipes-core/packagegroups/packagegroup-pipewire.bb +++ b/meta-pipewire/recipes-core/packagegroups/packagegroup-pipewire.bb @@ -12,4 +12,5 @@ RDEPENDS_${PN} += "\ agl-service-audiomixer \ pipewire \ pipewire-alsa \ + gstreamer1.0-pipewire \ " 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" |