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gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...

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Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc):
Preserve the existing buggy streamselector behaviour by performing
a fallback buffer allocation when downstream isn't linked yet.
This should really be fixed in playbin by blocking pads until it's
linked them.
Also, use gst_pad_alloc_buffer instead of
gst_pad_alloc_buffer_and_set.
This commit is contained in:
Jan Schmidt 2006-03-16 10:22:27 +00:00
parent c6e2d1eeea
commit 2caf3a5639
4 changed files with 129 additions and 14 deletions
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12
ChangeLog
View file

@ -1,3 +1,15 @@
2006-03-16 Jan Schmidt <thaytan@mad.scientist.com>
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc):
Preserve the existing buggy streamselector behaviour by performing
a fallback buffer allocation when downstream isn't linked yet.
This should really be fixed in playbin by blocking pads until it's
linked them.
Also, use gst_pad_alloc_buffer instead of
gst_pad_alloc_buffer_and_set.
2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/tag/gstid3tag.c:

111
ext/alsa/gstalsasink.c
View file

@ -117,18 +117,29 @@ static GstStaticPadTemplate alsasink_sink_factory =
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 32, "
"depth = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 8 ]; "
"rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]; "
"audio/x-raw-int, "
"endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 24, "
"depth = (int) { 24, 20, 18 }, "
"rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]; "
"audio/x-raw-int, "
"endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 8 ]; "
"rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 8 ]")
"rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]; "
"audio/x-alaw, rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]; "
"audio/x-mulaw, rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]; "
"audio/x-raw-float, width = (int) { 32, 64 }, "
"endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
"rate = (int) [ 1, MAX ], channels = (int) [ 1, 8 ]")
);
static GstElementClass *parent_class = NULL;
@ -312,6 +323,56 @@ get_channel_free_structure (const GstStructure * in_structure)
return s;
}
/* Get the GStreamer caps for a given Alsa format. NULL if we
* don't have a mapping */
static GstCaps *
gst_alsa_get_caps_for_format (snd_pcm_format_t format)
{
if (format == SND_PCM_FORMAT_A_LAW) {
return gst_caps_new_simple ("audio/x-alaw", NULL);
} else if (format == SND_PCM_FORMAT_MU_LAW) {
return gst_caps_new_simple ("audio/x-mulaw", NULL);
} else if (snd_pcm_format_linear (format)) {
/* int */
GstStructure *structure = gst_structure_new ("audio/x-raw-int",
"width", G_TYPE_INT, (gint) snd_pcm_format_physical_width (format),
"depth", G_TYPE_INT, (gint) snd_pcm_format_width (format),
"signed", G_TYPE_BOOLEAN,
snd_pcm_format_signed (format) == 1 ? TRUE : FALSE,
NULL);
/* endianness */
if (snd_pcm_format_physical_width (format) > 8) {
switch (snd_pcm_format_little_endian (format)) {
case 0:
gst_structure_set (structure, "endianness", G_TYPE_INT, G_BIG_ENDIAN,
NULL);
break;
case 1:
gst_structure_set (structure, "endianness", G_TYPE_INT,
G_LITTLE_ENDIAN, NULL);
break;
default:
GST_WARNING
("Unknown byte order in sound driver. Continuing by assuming system byte order.");
gst_structure_set (structure, "endianness", G_TYPE_INT, G_BYTE_ORDER,
NULL);
break;
}
}
return gst_caps_new_full (structure, NULL);
} else if (snd_pcm_format_float (format)) {
/* no float with non-platform endianness */
if (!snd_pcm_format_cpu_endian (format))
return NULL;
return gst_caps_new_simple ("audio/x-raw-float",
"width", G_TYPE_INT, (gint) snd_pcm_format_width (format),
"endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
}
return NULL;
}
static void
caps_add_channel_configuration (GstCaps * caps,
const GstStructure * in_structure, gint min_channels, gint max_channels)
@ -384,7 +445,7 @@ gst_alsasink_getcaps (GstBaseSink * bsink)
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstAlsaSink *sink = GST_ALSA_SINK (bsink);
GstCaps *tmpl_caps;
GstCaps *avail_caps;
GstCaps *caps = NULL;
guint min, max;
gint i, err, min_channels, max_channels;
@ -433,23 +494,53 @@ gst_alsasink_getcaps (GstBaseSink * bsink)
GST_LOG_OBJECT (sink, "Min. channels = %d (%d)", min_channels, min);
GST_LOG_OBJECT (sink, "Max. channels = %d (%d)", max_channels, max);
/* Get the list of formats that this sound device supports and
* generate caps from it */
avail_caps = gst_caps_new_empty ();
snd_pcm_format_mask_alloca (&mask);
snd_pcm_hw_params_get_format_mask (hw_params, mask);
element_class = GST_ELEMENT_GET_CLASS (sink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
for (i = 0; i <= SND_PCM_FORMAT_LAST; i++) {
if (snd_pcm_format_mask_test (mask, i)) {
GstCaps *cur_caps = gst_alsa_get_caps_for_format (i);
g_return_val_if_fail (pad_template != NULL, NULL);
GST_DEBUG_OBJECT (sink, "supported format: %02d %s", i,
snd_pcm_format_name (i));
tmpl_caps = gst_pad_template_get_caps (pad_template);
if (cur_caps != NULL)
gst_caps_append (avail_caps, cur_caps);
}
}
/* Template caps represents the caps we want to allow, intersect those
with what the sound device supports */
{
GstCaps *tmp_caps;
element_class = GST_ELEMENT_GET_CLASS (sink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
tmp_caps = gst_caps_intersect (avail_caps,
gst_pad_template_get_caps (pad_template));
gst_caps_unref (avail_caps);
avail_caps = tmp_caps;
}
GST_LOG_OBJECT (sink, "Sound device supports formats %" GST_PTR_FORMAT,
avail_caps);
/* Now prepare the caps we'll return */
caps = gst_caps_new_empty ();
for (i = 0; i < gst_caps_get_size (tmpl_caps); ++i) {
for (i = 0; i < gst_caps_get_size (avail_caps); ++i) {
caps_add_channel_configuration (caps,
gst_caps_get_structure (tmpl_caps, i), min_channels, max_channels);
gst_caps_get_structure (avail_caps, i), min_channels, max_channels);
}
gst_caps_unref (avail_caps);
sink->cached_caps = gst_caps_ref (caps);
GST_DEBUG_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);

16
gst/playback/gststreamselector.c
View file

@ -306,8 +306,20 @@ gst_stream_selector_bufferalloc (GstPad * pad, guint64 offset,
result = GST_FLOW_NOT_LINKED;
} else {
result = gst_pad_alloc_buffer_and_set_caps (sel->srcpad, offset,
size, caps, buf);
result = gst_pad_alloc_buffer (sel->srcpad, offset, size, caps, buf);
/* FIXME: HACK. If buffer alloc returns not-linked, perform a fallback
* allocation. This should NOT be necessary, because playbin should
* properly block the source pad from running until it's finished hooking
* everything up, but playbin needs refactoring first. */
if (result == GST_FLOW_NOT_LINKED) {
GST_DEBUG_OBJECT (sel,
"No peer pad yet - performing fallback allocation for pad %s:%s",
GST_DEBUG_PAD_NAME (pad));
*buf = NULL;
result = GST_FLOW_OK;
}
}
gst_object_unref (sel);

4
win32/common/config.h
View file

@ -42,7 +42,7 @@
#define GST_PACKAGE_ORIGIN "Unknown package origin"
/* package name in plugins */
#define GST_PACKAGE_NAME "GStreamer Base Plug-ins source release"
#define GST_PACKAGE_NAME "GStreamer Base Plug-ins CVS/prerelease"
/* Define the version */
#define GST_VERSION "@GST_VERSION@"
@ -250,7 +250,7 @@
#undef STDC_HEADERS
/* Version number of package */
#define VERSION "0.10.5"
#define VERSION "0.10.5.1"
/* Define to 1 if your processor stores words with the most significant byte
first (like Motorola and SPARC, unlike Intel and VAX). */