Commit graph

18927 commits

Author SHA1 Message Date
Seungha Yang
2ef74f2c81 qtmux: Fix incompatible type warning with MSVC
gstqtmux.c(5582): warning C4133: 'function':
  incompatible types - from 'GstVideoMultiviewFlags *' to 'guint *'
2019-09-02 15:07:17 +00:00
Mathieu Duponchelle
c5e8a8f320 rtspsrc: fix git diff indentation 2019-09-02 16:33:05 +02:00
Mathieu Duponchelle
3bc5d3d3b5 rtspsrc: normalize variable to boolean 2019-08-30 22:42:58 +02:00
Mathieu Duponchelle
37eca8a12c rtspsrc: clip output segment on accurate seeks
The output segment is only used in ONVIF mode.

The previous behaviour was to output a segment computed from
the Range response sent by the server.

In ONVIF mode, servers will start serving from the appropriate
synchronization point (keyframe), and the Range in response will
start at that position.

This means rtspsrc can now perform truly accurate seeks in that
mode, by clipping the output segment to the values requested in
the seek. The decoder will then discard out of segment buffers
and playback will start without artefacts at the exact requested
position, similar to the behaviour of a demuxer when an accurate
seek is requested.
2019-08-30 14:50:21 +00:00
Matthew Waters
0017115494 vpx: fix macos werror build
../ext/vpx/gstvpxenc.c:1723:49: error: format specifies type 'long' but the argument has type 'vpx_codec_pts_t' (aka 'long long') [-Werror,-Wformat]
          ", gst frame pts: %" G_GINT64_FORMAT, pkt->data.frame.pts, pts);
          ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
/Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:1065:96: note: expanded from macro 'GST_TRACE_OBJECT'
#define GST_TRACE_OBJECT(obj,...)       GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_TRACE,   obj,  __VA_ARGS__)
                                        ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~
/Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:646:31: note: expanded from macro 'GST_CAT_LEVEL_LOG'
        (GObject *) (object), __VA_ARGS__);                             \
                              ^~~~~~~~~~~
../ext/vpx/gstvpxenc.c:1723:70: error: format specifies type 'long' but the argument has type 'vpx_codec_pts_t' (aka 'long long') [-Werror,-Wformat]
          ", gst frame pts: %" G_GINT64_FORMAT, pkt->data.frame.pts, pts);
          ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~
/Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:1065:96: note: expanded from macro 'GST_TRACE_OBJECT'
#define GST_TRACE_OBJECT(obj,...)       GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_TRACE,   obj,  __VA_ARGS__)
                                        ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~
/Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:646:31: note: expanded from macro 'GST_CAT_LEVEL_LOG'
        (GObject *) (object), __VA_ARGS__);                             \
                              ^~~~~~~~~~~
2019-08-30 14:12:29 +10:00
Matthew Waters
b0522aa8f7 osxvideosink: call superclass in reshape
Fixes macos werror build

../sys/osxvideo/cocoawindow.m:437:1: error: method possibly missing a [super reshape] call [-Werror,-Wobjc-missing-super-calls]
}
^
2019-08-30 14:12:29 +10:00
Mathieu Duponchelle
3429ddde38 docstrings: port ulinks to markdown links 2019-08-23 18:56:01 +02:00
Tim-Philipp Müller
0dc9e5bff8 replaygain: fix up doc links to defunct replaygain.org website
Fixes #624
2019-08-23 13:12:39 +03:00
Seungha Yang
53dc1da870 souphttpsrc: Fix incompatible type build warning
gstsouphttpsrc.c(2191): warning C4133:
  '=': incompatible types - from 'guint (__cdecl *)(GType)' to 'GstURIType (__cdecl *)(GType)'
2019-08-21 15:20:56 +00:00
Tim-Philipp Müller
61bd613de3 vpx: bump libvpx requirement to 1.5.0
Was released in Nov 2015.
2019-08-19 11:09:47 +01:00
Tim-Philipp Müller
e40ba71824 vpx: avoid confusing meson configure output when checking for vpx versions
Used to print:
|Run-time dependency vpx found: YES 1.7.0
|Message: libvpx provides VP8 encoder interface (vpx_codec_vp8_cx_algo)
|Message: libvpx provides VP8 decoder interface (vpx_codec_vp8_dx_algo)
|Message: libvpx provides VP9 encoder interface (vpx_codec_vp9_cx_algo)
|Message: libvpx provides VP9 decoder interface (vpx_codec_vp9_dx_algo)
|Dependency vpx found: YES (cached)
|Dependency vpx found: NO found '1.7.0' but need: '>=1.8.0'
|Run-time dependency vpx found: NO (tried pkgconfig and cmake)

We can check the version of the found dep in a way that
doesn't produce this confusing output.
2019-08-19 11:09:19 +01:00
Amr Mahdi
cbe61c4ff5 wavparse: Fix push mode ignoring audio with a size smaller than segment buffer
In push mode (streaming), if the audio size is smaller than segment buffer size, it would be ignored.
This happens because when the plugin receives an EOS signal while a single audio chunk that is less than the segment buffer size is buffered, it does not
flush this chunk. The fix is to flush the data chunk when it receives an EOS signal and has a single (first) chunk buffered.

How to reproduce:
1. Run gst-launch with tcp source
```
gst-launch-1.0  tcpserversrc port=3000 !  wavparse ignore-length=0 ! audioconvert ! filesink location=bug.wav
```
2. Send a wav file with unspecified data chunk length (0). Attached a test file
```
cat test.wav | nc localhost 3000
```
3. Compare the length of the source file and output file
```
ls -l test.wav bug.wav
-rw-rw-r-- 1 amr amr    0 Aug 15 11:07 bug.wav
-rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav
```

The expected length of the result of the gst-lauch pipeline should be the same as the test file minus the headers (44), which is ```3564 - 44 = 3520``` but the actual output length is ```0```

After the fix:
```
ls -l test.wav fix.wav
-rw-rw-r-- 1 amr amr 3520 Aug 15 11:09 fix.wav
-rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav
```
2019-08-19 07:30:17 +00:00
Sebastian Dröge
2a4d0a9b09 rtpvp8depay: Add property for waiting until the next keyframe after packet loss
If VP8 is not encoded with error resilience enabled then any packet loss
causes very bad artefacts when decoding and waiting for the next
keyframe instead improves user experience considerably.
2019-08-12 17:10:20 +00:00
Nicolas Dufresne
45b6514f33 v4l2: Fix type compatibility issue with glibc 2.30
From now on, we will use linux/types.h on Linux, and use typedef of the
various flavour of BSD.

Fixes #635
2019-08-10 11:01:26 +01:00
Mathieu Duponchelle
e58ca79741 valgrind: suppress Cond error coming from gnutls
taken from fb4a8dda21
2019-08-08 14:39:17 +00:00
Mart Raudsepp
67958ccce8 matroska: Provide audio lead-in for some lossy formats
Various audio formats require an audio lead-in to decode it properly.
Most parsers would take care of it, but when a container like matroska is
involved, the demuxer handles the seeking and without its own lead-in
handling would never even pass the lead-in data to the parser.
This commit provides an initial implementation of that for audio/mpeg,
audio/x-ac3 and audio/x-eac3 by calculating the worst case lead-in time
needed from known samplerate, potential lead-in frames need and the
maximum blocksize possible for the format (as we don't parse that out
exactly in matroskademux) and seeking that much earlier in case of
accurate seeks. This is especially important for NLE use-cases with GES.

If accurate seeking to a position that happens to have a video keyframe,
it'll go back to the previous keyframe than needed, but with typical
video files that's the best we can do anyway without falling back to
scanning the clusters, as typically only keyframes are indexed in
Cueing Data.
If the media doesn't have a CUE, then we bisect for the cluster to seek
to with the same modified time as well in case of accurate seeking,
ensuring sufficient lead-in. This code path is typically hit only with
(suboptimal) audio-only matroska files, e.g. when created with ffmpeg,
which doesn't add a CUE for audio-only mkv muxing.
2019-08-07 18:51:57 -04:00
Antonio Ospite
1c5c90ea23 test: rtpbin_buffer_list: add a test for invalid packets in buffer list
Upstream elements can send all kinds of data in a buffer list, so cover
the case of an invalid RTP packet mixed with valid RTP packets.
2019-08-07 15:32:30 -04:00
Antonio Ospite
12b420168c test: rtpbin_buffer_list: add a test for multiplexed RTP and RTCP
RTP and RTCP packets can be muxed together on the same channel (see
RFC5761) and can arrive in the same buffer list.

The GStreamer rtpsession element support RFC5761, so add a test to cover
this case for buffer lists too.
2019-08-07 15:32:30 -04:00
Antonio Ospite
6a5f38a325 test: rtpbin_buffer_list: add a test for different timestamps in buffer list
Buffers with different timestamps (e.g. packets belonging to different
frames) can arrive together in the same buffer list,

Add a test to cover this case.
2019-08-07 15:32:30 -04:00
Antonio Ospite
b158be0d98 test: rtpbin_buffer_list: add function to check timestamp 2019-08-07 15:32:30 -04:00
Antonio Ospite
31f221f89d test: rtpbin_buffer_list: add a test about reordered or duplicated seqnums 2019-08-07 15:32:30 -04:00
Antonio Ospite
1d337b704e test: rtpbin_buffer_list: add a test for lange jump in seqnums with recovery 2019-08-07 15:32:30 -04:00
Antonio Ospite
ae5d4b8cf0 test: rtpbin_buffer_list: add a test for large jump in sequence numbers 2019-08-07 15:32:30 -04:00
Antonio Ospite
fb46c6bf08 test: rtpbin_buffer_list: add a test for wrapping sequence numbers 2019-08-07 15:32:30 -04:00
Antonio Ospite
d9d0461aeb test: rtpbin_buffer_list: add a test for permissible gap in sequence numbers 2019-08-07 15:32:30 -04:00
Antonio Ospite
92be4121e8 test: rtpbin_buffer_list: add a test for the case of failed probation
When a new source fails to pass the probation period (i.e. new packets
have non-consecutive sequence numbers), then no buffer shall be pushed
downstream. Add a test to validate this case.
2019-08-07 15:32:30 -04:00
Antonio Ospite
0537925c3a test: rtpbin_buffer_list: add function to check sequence number 2019-08-07 15:32:30 -04:00
Antonio Ospite
ec56e2fb78 test: rtpbin_buffer_list: add test to verify that receiving stats are correct
Add a test to verify that stats about received packets are correct when
using buffer lists in the rtpsession receive path.

Split get_session_source_stats() in two to be able to get stats from
a GstRtpSession object directly.
2019-08-07 15:32:30 -04:00
Antonio Ospite
80daea90f0 test: rtpbin_buffer_list: add a test for buffer lists on the recv path 2019-08-07 15:32:30 -04:00
Antonio Ospite
8dd03042cc rtpsession: add support for buffer lists on the recv path
The send path in rtpsession processes the buffer list along the way,
sharing info and stats between packets in the same list, because it
assumes that all packets in a buffer list are from the same frame.

However, in the receiving path packets can arrive in all sorts of
arrangements:

  - different sources,
  - different frames (different timestamps),
  - different types (multiplexed RTP and RTCP, invalid RTP packets).

so a more general approach should be used to correctly support buffer
lists in the receive path.

It turns out that it's simpler and more robust to process buffers
individually inside the rtpsession element even if they come in a buffer
list, and then reassemble a new buffer list when pushing the buffers
downstream.

This avoids complicating the existing code to make all functions
buffer-list-aware with the risk of introducing regressions,

To support buffer lists in the receive path and reduce the "push
overhead" in the pipeline, a new private field named processed_list is
added to GstRtpSessionPrivate, it is set in the chain_list handler and
used in the process_rtp callback; this is to achieve the following:

  - iterate over the incoming buffer list;
  - process the packets one by one;
  - add the valid ones to a new buffer list;
  - push the new buffer list downstream.

The processed_list field is reset before pushing a buffer list to be on
the safe side in case a single buffer was to be pushed by upstream
at some later point.

NOTE:

The proposed modifications do not change the behavior of the send path.

The process_rtp callback is called in rtpsource.c by the push_rtp
callback (via source_push_rtp) only when the source is not internal.

So even though push_rtp is also called in the send path, it won't end up
using process_rtp in this case because the source would be internal in
the send path.

The reasoning from above may suggest a future refactoring: push_rtp
might be split to better differentiate the send and receive path.
2019-08-07 15:32:30 -04:00
Doug Nazar
b0534c65d1 matroska: Handle interlaced field order 2019-08-07 14:12:32 +00:00
Amr Mahdi
5f01b9da05 wavparse: Fix ignoring of last chunk in push mode
In push mode (streaming), if the last audio payload chunk is less than the segment rate buffer size, it would be ignored since the plugin waits until it has at least segment rate bufer size of audio.

The fix is to introduce a flushing flag that indicates that no more audio will be available so that the plugin can recognize this condition and flush the data is has even if it is less
than the desired segment rate buffer size.
2019-08-07 12:09:46 +00:00
Robert Tiemann
3c4add469e souphttpsrc: Log any error returned by soup_session_send() 2019-08-07 07:32:09 +00:00
luke.lin
d6ae59c32d qtdemux: enlarge the maximal atom size
For 8K content, frame size is over 25MB, and cause the negotiation failure.
Enlarge the limitation of QTDEMUX_MAX_ATOM_SIZE to 32MB.
2019-08-07 02:46:20 +00:00
Mathieu Duponchelle
5c7423d73c rtspsrc: expose and implement is-live property
This is useful to support the ONVIF case: when is-live is set to
FALSE and onvif-rate-control is no, the client can control the
rate of delivery and arrange for the server to block and still
keep sending when unblocked, without requiring back and forth
PAUSE / PLAY requests. This enables, amongst other things, fast
frame stepping on the client side.

When is-live is FALSE, we don't use a manager at all. This case
was actually already pretty well handled by the current code. The
standard manager, rtpbin, is simply no longer needed in this case.

Applications can instantiate a downloadbuffer after rtspsrc if
needed.
2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
75f53631e5 rtspsrc: reset_time when flush stopping 2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
5f1a732bc7 rtspsrc: expose and implement onvif-mode property
Refactor the code for parsing and generating the Range, taking
advantage of existing API in GstRtspTimeRange.

Only use the TCP protocol in that mode, as per the specification.

Generate an accurate segment when in that mode, and signal to the
depayloader that it should not generate its own segment, through
the "onvif-mode" field in the caps, see
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/328>
for more information.

Translate trickmode seek flags to their ONVIF representation

Expose an onvif-rate-control property
2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
544f8fecf4 rtspsrc: improve handling of rate in seeks 2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
e18d5d6ec6 rtpfunnel: forward correct segment when switching pad
Forwarding a single segment event from the pad that first gets
chained is incorrect: when that first event was sent by an element
such as x264enc, with its offset start, we end pushing out of segment
buffers for the other pad(s).

Instead, everytime the active pad changes, forward the appropriate
segment event.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1028
2019-08-06 14:02:50 +00:00
Sebastian Dröge
86ec5c1031 rtspsrc: Use new GstRTSPMessage API to set message body from a buffer directly 2019-08-05 19:35:36 +03:00
Antonio Ospite
ae48646d8e rtpsource: fix receiver source stats to consider previously queued packets
When it is not clear yet if a packet relative to a source should be
pushed, the packet is put into a queue, this happens in two cases:

  - the source is still in probation;
  - there is a large jump in seqnum, and it is not clear what
    the cause is, future packets will help making a guess.

In either case stats about received packets are not updated at all; and
even if they were, when init_seq() is called it resets all receiver
stats, effectively loosing any possible stat about previously received
packets.

Fix this by taking into account the queued packets and update the stats
when calling init_seq().
2019-08-02 17:22:51 +02:00
Antonio Ospite
cf0ffd8693 rtpsource: clarify meaning of the octets-sent and octets-received stats
The octets-send and octets-received stats count the payload bytes
excluding RTP and lower level headers, clarify that in the
documentation.
2019-08-02 17:22:51 +02:00
Antonio Ospite
821994240e rtpsource: expose field bytes_received in RTPSourceStats
Since commit c971d1a9a (rtpsource: refactor bitrate estimation,
2010-03-02) bytes_received filed in RTPSourceStats is set but then never
used again, expose it so that it can be used  by user code to verify how
many bytes have been received.
2019-08-02 17:22:51 +02:00
Antonio Ospite
9d800cad43 rtpmanager: consider UDP and IP headers in bandwidth calculation
According to RFC3550 lower-level headers should be considered for
bandwidth calculation.

See https://tools.ietf.org/html/rfc3550#section-6.2 paragraph 4:

  Bandwidth calculations for control and data traffic include
  lower-layer transport and network protocols (e.g., UDP and IP) since
  that is what the resource reservation system would need to know.

Fix the source data to accommodate that.

Assume UDPv4 over IP for now, this is a simplification but it's good
enough for now.

While at it define a constant and use that instead of a magic number.

NOTE: this change basically reverts the logic of commit 529f443a6
(rtpsource: use payload size to estimate bitrate, 2010-03-02)
2019-08-02 17:22:51 +02:00
Seungha Yang
4146dc905d qtdemux: Use empty-array safe way to cleanup GPtrArray
Fix assertion fail
GLib-CRITICAL **: g_ptr_array_remove_range: assertion 'index_ < rarray->len' failed
2019-08-02 12:32:59 +09:00
Marc Leeman
d365c4fdf9 rtpmp4vpay: config-interval -1 send at idr
adjust/port from rtph264pay and allow sending the configuration data at
every IDR

The payloader was stripping the configuration data when the
config-interval was set to 0. The code was written in such a way !(a >
0) that it stripped the config when it was set at -1 (send config_data
as soon as possible).

This resulted in some MPEG4 streams where no GOP/VOP-I was detected to
be sent out without configuration.
2019-08-01 14:28:04 +00:00
Doug Nazar
5451e4e900 matroskademux: Ignore crc32 element while peeking at cluster. 2019-07-27 14:21:34 -04:00
Guillaume Desmottes
5c4292b019 gtkglsink: fix crash when widget is resized after element destruction
Prevent _size_changed_cb() to be called after gtkglsink has been finalized.

Fix #632
2019-07-26 08:25:45 +00:00
Mathieu Duponchelle
4830bbe6ca qtdemux: fix reverse playback EOS conditions
In reverse playback, we don't want to rely on the position of the current
keyframe to decide a stream is EOS: the last GOP we push will start with
a keyframe, which position is likely to be outside of the segment.

Instead, let the normal seek_to_previous_keyframe mechanism do its job,
it works just fine.
2019-07-26 02:42:11 +00:00
Mathieu Duponchelle
104f459258 qtdemux: fix key unit seek corner case
If a key unit seek is performed with a time position that matches
the offset of a keyframe, but not its actual PTS, we need to
adjust the segment nevertheless.

For example consider the following case:

* stream starts with a keyframe at 0 nanosecond, lasting 40 milliseconds
* user does a key unit seek at 20 milliseconds
* we don't adjust the segment as the time position is "over" a keyframe
* we push a segment that starts at 20 milliseconds
* we push a buffer with PTS == 0
* an element downstream (eg rtponviftimestamp) tries to calculate the
  stream time of the buffer, fails to do so and drops it
2019-07-26 01:50:47 +00:00