mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-02-17 11:45:25 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.

Browse source 
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
This commit is contained in:
Wim Taymans 2006-09-27 13:52:14 +00:00
parent bfbdfb8d45
commit 1980f16731
4 changed files with 38 additions and 9 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

17
ChangeLog
View file

@ -1,3 +1,20 @@
2006-09-27 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
* tests/check/Makefile.am:

1
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -649,6 +649,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
}
no_align:
/* crate contains diff against the clock we are using in the pipeline. */
crate =
gst_guint64_to_gdouble (crate_num) / gst_guint64_to_gdouble (crate_denom);
GST_DEBUG_OBJECT (sink,

22
gst-libs/gst/audio/gstbaseaudiosrc.c
View file

@ -449,10 +449,10 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer *buf;
guchar *data;
guint samples;
guint res;
guint64 sample;
gint bps;
GstRingBuffer *ringbuffer;
guint read;
ringbuffer = src->ringbuffer;
@ -488,10 +488,22 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
buf = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (buf);
/* read the sample */
res = gst_ring_buffer_read (ringbuffer, sample, data, samples);
if (G_UNLIKELY (res == -1))
goto stopped;
do {
read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
/* if we read all, we're done */
if (read == samples)
break;
/* else something interrupted us and we wait for playing again. */
if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
goto stopped;
/* read next samples */
sample += read;
samples -= read;
data += read * bps;
} while (TRUE);
/* mark discontinuity if needed */
if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {

7
gst-libs/gst/audio/gstringbuffer.c
View file

@ -134,8 +134,8 @@ gst_ring_buffer_finalize (GObject * object)
typedef struct
{
GstBufferFormat format;
guint8 silence[4];
const GstBufferFormat format;
const guint8 silence[4];
} FormatDef;
static const FormatDef linear_defs[4 * 2 * 2] = {
@ -1387,8 +1387,7 @@ gst_ring_buffer_read (GstRingBuffer * buf, guint64 sample, guchar * data,
not_started:
{
GST_DEBUG_OBJECT (buf, "stopped processing");
/* FIXME, return len - to_read after fixing caller */
return -1;
return len - to_read;
}
}