-From e2426512322768a2d386740845c80e282aba5a6a Mon Sep 17 00:00:00 2001
+From 1b1f884a165ed7b2147affbdddf85a641d4cf180 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
Both elements share a common ringbuffer that actually implements
the pipewire integration.
-Upstream-Status: Submitted [https://github.com/PipeWire/pipewire/pull/140]
+Upstream-Status: Denied
+See https://gitlab.freedesktop.org/pipewire/pipewire/merge_requests/140
---
src/gst/gstpipewire.c | 8 +-
- src/gst/gstpwaudioringbuffer.c | 542 +++++++++++++++++++++++++++++++++
+ src/gst/gstpwaudioringbuffer.c | 565 +++++++++++++++++++++++++++++++++
src/gst/gstpwaudioringbuffer.h | 83 +++++
- src/gst/gstpwaudiosink.c | 200 ++++++++++++
+ src/gst/gstpwaudiosink.c | 207 ++++++++++++
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(-)
+ 8 files changed, 1164 insertions(+), 1 deletion(-)
create mode 100644 src/gst/gstpwaudioringbuffer.c
create mode 100644 src/gst/gstpwaudioringbuffer.h
create mode 100644 src/gst/gstpwaudiosink.c
}
diff --git a/src/gst/gstpwaudioringbuffer.c b/src/gst/gstpwaudioringbuffer.c
new file mode 100644
-index 00000000..989b2cd7
+index 00000000..babf2d83
--- /dev/null
+++ b/src/gst/gstpwaudioringbuffer.c
-@@ -0,0 +1,542 @@
+@@ -0,0 +1,565 @@
+/* PipeWire
+ *
+ * Copyright © 2018 Wim Taymans
+ 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));
+ break;
+ 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);
+}
+
+static void
-+on_stream_format_changed (void *data, const struct spa_pod *format)
++on_stream_param_changed (void *data, uint32_t id, 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];
+
++ if (format == NULL || id != SPA_PARAM_Format)
++ return;
++
+ 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_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);
++ pw_stream_update_params (self->stream, params, 1);
+}
+
+static void
+
+ b = pw_stream_dequeue_buffer (self->stream);
+ if (!b) {
-+ GST_WARNING_OBJECT (self->elem, "no pipewire buffer available");
++ GST_INFO_OBJECT (self->elem, "no pipewire buffer available");
+ return;
+ }
+
+static const struct pw_stream_events stream_events = {
+ PW_VERSION_STREAM_EVENTS,
+ .state_changed = on_stream_state_changed,
-+ .format_changed = on_stream_format_changed,
++ .param_changed = on_stream_param_changed,
+ .process = on_stream_process,
+};
+
+
+ /* construct param & props objects */
+
++ props = pw_properties_new (NULL, NULL);
+ 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));
+ self->rate = GST_AUDIO_INFO_RATE (&spec->info);
+ self->segoffset = 0;
+
++ pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%u/%u",
++ self->segsize / self->bpf, self->rate);
++ GST_DEBUG_OBJECT (self->elem, "segsize:%u, bpf:%u, node.latency = %s",
++ self->segsize, self->bpf, pw_properties_get (props, PW_KEY_NODE_LATENCY));
++
+ /* connect stream */
+
+ pw_thread_loop_lock (self->main_loop);
+ 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_PAUSED))
+ goto start_error;
+
+ pw_thread_loop_unlock (self->main_loop);
+#endif
diff --git a/src/gst/gstpwaudiosink.c b/src/gst/gstpwaudiosink.c
new file mode 100644
-index 00000000..6cb71385
+index 00000000..069996c3
--- /dev/null
+++ b/src/gst/gstpwaudiosink.c
-@@ -0,0 +1,200 @@
+@@ -0,0 +1,207 @@
+/* PipeWire
+ *
+ * Copyright © 2018 Wim Taymans
+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
pipewire_gst_c_args = [
--
-2.23.0
+2.24.0
Code Review / AGL / meta-agl-devel.git / blobdiff