This allows us to output audio samples without discarding
any input frames, which is useful for some formats/codecs
(e.g. the MonkeysAudio decoder implementation in ffmpeg
which will might return e.g. 16 output buffers for an
input buffer for certain files).
In the past decoder implementations just concatenated
the returned audio buffers until a full frame had been
decoded, but that's no longer possible to do efficiently
when the decoder returns audio samples in non-interleaved
layout.
Allowing subframes to be output before the entire input
frame is decoded can also be useful to decrease startup
latency/delay.
https://gitlab.freedesktop.org/gstreamer/gst-libav/issues/49
../subprojects/gst-plugins-base/tests/check/elements/audiorate.c(192): warning C4047
Meaningful validation at that point seems to checking output GstAudioFormat
of gst_audio_format_from_string()
The use of mediump as a specifier in GLSL shaders will have limited
resolution and when used as texture coordinates may become inaccurate
over texture sizes of 1024.
This will only duplicate buffers if the gap between two consecutive
buffers is up to fill-until nsec. If it's larger, it will only output
the new buffer and mark it as discont.
New casts to avoid the the warnings mentioned below. While at it, move
some existing casts (introduced at 61bc909189) to use
GPOINTER_TO_INT too.
[458/673] Compiling C object 'tests/check/7d01337@@libs_video@exe/libs_video.c.obj'.
../tests/check/libs/video.c: In function 'fourcc_get_size':
../tests/check/libs/video.c:160:10: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
return (unsigned long) p->endptr;
^
In file included from ../tests/check/libs/video.c:32:
../tests/check/libs/video.c: In function 'test_video_formats':
../tests/check/libs/video.c:563:39: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
fail_unless_equals_int (size, (unsigned long) paintinfo.endptr);
^
And more.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/94
With commit 3f184c3abc, the gst_dir variable becomes unusable in
windows build. Moving it to linux scope to avoid warning:
[433/673] Compiling C object 'tests/check/7d01337@@libs_profile@exe/libs_profile.c.obj'.
../tests/check/libs/profile.c: In function 'profile_suite':
../tests/check/libs/profile.c:688:10: warning: unused variable 'gst_dir' [-Wunused-variable]
gchar *gst_dir;
^~~~~~~
Also fix a typo in the comment.
The function fill_bytes could sometimes return a value greater than zero
and in the same time set the GError.
Function read_bytes calls fill_bytes in a while loop. In the special
case above it would call fill_bytes with error already set.
Thus resulting in "GError set over the top of a previous GError".
Solved this by clearing GError when return value is greater than zero.
Actions are taken depending on error type by caller of read_bytes. Eg.
with EWOULDBLOCK gst_rtsp_source_dispatch_read will try to read the
missing bytes again (GST_RTSP_EINTR )
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/445
It is really easy to break the API and insert a new video format in the
middle of the enum instead of at the end. This minimal test should catch
the most obvious errors. Ideally, this test should be updated after new
format have been added, so that it won't allow further modification to
the enumeration API.
rtpbasedepayload.c:126:5: error: unknown conversion type character 'z' in format [-Werror=format]
profile.c:688:10: error: unused variable 'gst_dir' [-Werror=unused-variable]
gst_video_decoder_negotiate_default_caps() is meant to pick a default output
format when we need one earlier because of an incoming GAP.
It tries to use the input caps as a base if available and fallback to a default
format (I420 1280x720@30) for the missing fields.
But the framerate and pixel-aspect were not explicitly passed to
gst_video_decoder_set_output_state() which is solely relying on the input format
as reference to get the framerate anx pixel-aspect-ratio.
So there is no need to manually handling those two fields as
gst_video_decoder_set_output_state() will already use the ones from
upstream if available, and they will be ignored anyway if there are not.
This also prevent confusing debugging output where we claim to use a
specific framerate while actually none was set.
gstrtspconnection.c: In function ‘writev_bytes’:
gstrtspconnection.c:1348:10: error: ‘res’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
return res;
^
Allow fallback to orc subproject if any.
Additionally 'dependencies' keyword is removed from find_library,
because it's invalid keyword for find_library.
When using multichannel audio data and being needed to reorder channels,
audio data is not copied correctly because destination address of
memcpy is wrong.
For example, the following command
$ gst-launch-1.0 pulsesrc ! audio/x-raw,channels=6,format=S16LE ! filesink location=test.raw
will reproduce this issue if there is 6-ch audio input device.
This commit fixes that.
The detailed process of this issue is as follows:
1. gst-launch-1.0 calls gst_pulsesrc_prepare (gst-plugins-good/ext/pulse/pulsesrc.c)
1466 gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
1467 {
(skip...)
1480 {
1481 GstAudioRingBufferSpec s = *spec;
1482 const pa_channel_map *m;
1483
1484 m = pa_stream_get_channel_map (pulsesrc->stream);
1485 gst_pulse_channel_map_to_gst (m, &s);
1486 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
1487 (pulsesrc)->ringbuffer, s.info.position);
1488 }
In my environment, after line 1485 is processed, position of spec and s are
spec->info.position[0] = 0
spec->info.position[1] = 1
spec->info.position[2] = 2
spec->info.position[3] = 6
spec->info.position[4] = 7
spec->info.position[5] = 8
s.info.position[0] = 0
s.info.position[1] = 6
s.info.position[2] = 2
s.info.position[3] = 1
s.info.position[4] = 7
s.info.position[5] = 8
The values of spec->info.positions equal
GST_AUDIO_BASE_SRC(pulsesrc)->ringbuffer->spec->info.positions.
2. gst_audio_ring_buffer_set_channel_positions calls
gst_audio_get_channel_reorder_map.
3. Arguments of gst_audio_get_channel_reorder_map are
from = s.info.position
to = GST_AUDIO_BASE_SRC(pulsesrc)->ringbuffer->spec->info.positions
At the end of this function, reorder_map is set to
reorder_map[0] = 0
reorder_map[1] = 3
reorder_map[2] = 2
reorder_map[3] = 1
reorder_map[4] = 4
reorder_map[5] = 5
4. Go back to gst_audio_ring_buffer_set_channel_positions and
2065 buf->need_reorder = TRUE;
is processed.
5. Finally, in gst_audio_ring_buffer_read,
1821 if (need_reorder) {
(skip...)
1829 memcpy (data + i * bpf + reorder_map[j] * bps, ptr + j * bps, bps);
is processed and makes this issue.
Otherwise we would return EOF if nothing was written in any case, even
if this was actually a case of TIMEOUT or EWOULDBLOCK for example.
Thanks to Edward Hervey for debugging and finding this issue.
Fixes 2 problems:
1) Number of unmapped memories does not always match number of mmaped ones in
dispatch_write().
2) When dispatch_write() is dispatched second time after an incomplete write,
already set offsets will not be taken into account, thus corrupt RTP data will
be sent.
This makes it unnecessary for callers to first merge together all
memories, and it allows API like GstRTSPConnection to write them out
without first copying all memories together or using writev()-style API
to write multiple memories out in one go.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/370
g_source_remove() works only for a GSource which was attached
to default GMainContext, but the GSource might be attached to
custom context depending on how gst_discoverer_start() was called.
Whatever the attached context was, g_source_destroy() can clean it up.
