-From 8d448cc2d3bd141742ede4636d2208f4dedc16f6 Mon Sep 17 00:00:00 2001
+From 5a249321aa84cd74e3d83bcd555c85fba3cd682d Mon Sep 17 00:00:00 2001
From: George Kiagiadakis <george.kiagiadakis@collabora.com>
Date: Sun, 22 Sep 2019 17:59:19 +0300
Subject: [PATCH] meson: revert version check to require meson 0.47, not 0.50
-From c55e7b8e0028686afb74f4444b88c9e1dd6605c9 Mon Sep 17 00:00:00 2001
+From e4b81946baf2d8c08de87088c01a1d87ae4f03d9 Mon Sep 17 00:00:00 2001
From: George Kiagiadakis <george.kiagiadakis@collabora.com>
Date: Mon, 24 Jun 2019 12:19:20 +0300
Subject: [PATCH] arm: build with -mno-unaligned-access
-From 4696c0071d7a713d3e415f7d0cf29364a2b19aa2 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
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
-From c4028fd5d07bd6f901ca76cd7218150c658193f8 Mon Sep 17 00:00:00 2001
+From ce155eb0073fba84556782633f79bb7d03492c07 Mon Sep 17 00:00:00 2001
From: George Kiagiadakis <george.kiagiadakis@collabora.com>
Date: Wed, 2 Oct 2019 21:40:34 +0300
Subject: [PATCH] audioconvert: always assume that output ports are NOT monitor
+++ /dev/null
-From 1027efb2ced95a9548007ccc2f07805d48073846 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: Denied
-See https://gitlab.freedesktop.org/pipewire/pipewire/merge_requests/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 989b2cd7..97350f38 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);
---
-2.24.0
-
+++ /dev/null
-From ea2ba0fcf561fecd7e62e80e2c34bf6bd23f917b 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: Denied
-See https://gitlab.freedesktop.org/pipewire/pipewire/merge_requests/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 97350f38..3efec6ec 100644
---- a/src/gst/gstpwaudioringbuffer.c
-+++ b/src/gst/gstpwaudioringbuffer.c
-@@ -442,9 +442,9 @@ 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);
---
-2.24.0
-
-From 7bde580db9b67c503de90211d344b15b933fecbe Mon Sep 17 00:00:00 2001
+From 19fad1a4fa8bdc4f02aac4e169e7ff9cab18bdcd Mon Sep 17 00:00:00 2001
From: George Kiagiadakis <george.kiagiadakis@collabora.com>
Date: Tue, 19 Nov 2019 17:09:07 +0200
Subject: [PATCH] module-access: add same-sec-label-mode
+++ /dev/null
-From cc5ad0de4afb4c283e3c6b373b5fd891f9c19381 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: Denied
-See https://gitlab.freedesktop.org/pipewire/pipewire/merge_requests/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
---
-2.24.0
-
+++ /dev/null
-From 3357a7d5bfb32a26f6c4fd3df2ea3a8f04f20770 Mon Sep 17 00:00:00 2001
-From: George Kiagiadakis <george.kiagiadakis@collabora.com>
-Date: Tue, 20 Aug 2019 18:33:35 +0300
-Subject: [PATCH] gst: pwaudioringbuffer: set node.latency to get scheduled
- correctly in capture mode
-
-Upstream-Status: Denied
-See https://gitlab.freedesktop.org/pipewire/pipewire/merge_requests/140
----
- src/gst/gstpwaudioringbuffer.c | 7 ++++---
- 1 file changed, 4 insertions(+), 3 deletions(-)
-
-diff --git a/src/gst/gstpwaudioringbuffer.c b/src/gst/gstpwaudioringbuffer.c
-index 3efec6ec..8136b815 100644
---- a/src/gst/gstpwaudioringbuffer.c
-+++ b/src/gst/gstpwaudioringbuffer.c
-@@ -402,11 +402,9 @@ gst_pw_audio_ring_buffer_acquire (GstAudioRingBuffer *buf,
-
- /* 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));
-@@ -423,6 +421,9 @@ gst_pw_audio_ring_buffer_acquire (GstAudioRingBuffer *buf,
- 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);
-+
- /* connect stream */
-
- pw_thread_loop_lock (self->main_loop);
---
-2.24.0
-
file://0001-meson-revert-version-check-to-require-meson-0.47-not.patch \
file://0002-arm-build-with-mno-unaligned-access.patch \
file://0003-gst-Implement-new-pwaudio-src-sink-elements-based-on.patch \
- file://0004-gst-pwaudioringbuffer-request-pause-play-on-the-appr.patch \
- file://0005-gst-pwaudioringbuffer-wait-only-for-STREAM_STATE_CON.patch \
- file://0006-gst-pwaudiosink-set-the-default-latency-time-buffer-.patch \
- file://0007-gst-pwaudioringbuffer-set-node.latency-to-get-schedu.patch \
- file://0008-audioconvert-always-assume-that-output-ports-are-NOT.patch \
- file://0009-module-access-add-same-sec-label-mode.patch \
+ file://0004-audioconvert-always-assume-that-output-ports-are-NOT.patch \
+ file://0005-module-access-add-same-sec-label-mode.patch \
"
-SRCREV = "e18a24493a254c881a1bda384fdcd70cd671fd1c"
+SRCREV = "b0932e687fc47e0872ca291531f2291d99042d70"
PV = "0.2.91+git${SRCPV}+1"
S = "${WORKDIR}/git"
Code Review / AGL / meta-agl-devel.git / commitdiff