Android 19 added an API for dynamically changing the bitrate in a running
codec.
Also make it so that even when not update-able at runtime, parameters will at least
be stored so that they take effect the next the codec is restarted.
Change how content-length is set for HTTP POST headers, letting curl set
the header (given the content-length) instead of manually writing it.
This enables curl to know the content-length of the data.
In curl 7.66, if curl does not know the content-length (e.g. when
manually writing the header) curl will use Transfer-Encoding: chunked,
which might not be desired.
Use a double instead of a plain float for intermediary
property values, so we have enough bits to store INT_MAX
and it doesn't get rounded and wrapped to -1 when cast
back to a 32-bit integer.
Fixes criticals like
g_param_spec_int: assertion 'default_value >= minimum && default_value <= maximum' failed
when loading LADSPA plugins from the Linux Studio Plugins
Project (http://lsp-plug.in) in GStreamer.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1194
The avtpsink element is expected to transmit AVTPDUs at specific times,
according to GstBuffer timestamps. Currently, the transmission time is
controlled in software via the rendering synchronization mechanism
provided by GstBaseSink class. However, that mechanism may not cope with
some AVB use-cases such as Class A streams, where AVTPDUs are expected
to be transmitted at every 125 us. Thus, this patch introduces avtpsink
own mechanism which leverages the socket transmission scheduling
infrastructure introduced in Linux kernel 4.19. When supported by the
NIC, the transmission scheduling is offloaded to the hardware, improving
transmission time accuracy considerably.
To illustrate that, a before-after experiment was carried out. The
experimental setup consisted in 2 PCs with Intel i210 card connected
back-to-back running an up-to-date Archlinux with kernel 5.3.1. In one
host gst-launch-1.0 was used to generate a 2-minute Class A stream while
the other host captured the packets. The metric under evaluation is the
transmission interval and it is measured by checking the 'time_delta'
information from ethernet frames captured at the receiving side.
The table below shows the outcome for a 48 kHz, 16-bit sample, stereo
audio stream. The unit is nanoseconds.
| Mean | Stdev | Min | Max | Range |
-------+--------+---------+---------+---------+---------+
Before | 125000 │ 2401 │ 110056 │ 288432 │ 178376 |
After | 125000 │ 18 │ 124943 │ 125055 │ 112 |
Before this patch, the transmission interval mean is equal to the
optimal value (Class A stream -> 125 us interval), and it is kept the
same after the patch. The dispersion measurements, however, had
improved considerably, meaning the system is now consistently
transmitting AVTPDUs at the correct time.
Finally, the socket transmission scheduling infrastructure requires the
system clock to be synchronized with PTP clock so this patches modifies
the AVTP plugin documentation to cover how to achieve that.
This patch refactors gst_avtp_sink_start() by moving all socket
initialization code to its own function. This change prepares the code
to the next patch which will introduce avtpsink's own rendering
synchronization mechanism.
Current avtpsink code opens the AF_PACKET socket with SOCK_NONBLOCK
option. However, we actually want sendto() to block in case there isn't
available space in socket buffer.
This patch refactors both avtpsink and avtpsrc code so we use the
if_nametoindex() helper instead of building a request and issuing an
ioctl to get the if_index.
d3d11window holds one buffer to redraw client area per resize event.
When the input format is being changed, this buffer should be cleared
to avoid mismatch beween newly configured shader/videoprocessor and
the format of previously cached buffer.
Because the size of texture array cannot be updated dynamically,
allocator should block the allocation request. This cannot be
done at buffer pool side if this d3d11 memory is shared among
multiple buffer objects. Note that setting NO_SHARE flag to
d3d11 memory is very inefficient. It would cause most likey
copy of the d3d11 texture.
...for color space conversion if available
ID3D11VideoProcessor is equivalent to DXVA-HD video processor
which might use specialized blocks for video processing
instead of general GPU resource. In addition to that feature,
we need to use this API for color space conversion of DXVA2 decoder
output memory, because any d3d11 texture arrays that were
created with D3D11_BIND_DECODER cannot be used for shader resource.
This is prework for d3d11decoder zero-copy rendering and also
for conditional HDR tone-map support.
Note that some Intel platform is known to support tone-mapping
at the driver level using this API on Windows 10.
We've been using NvEncodeAPICreateInstance method to find the supported API
version, but that seems to be insufficient since there is a case
where plugin failed in opening encoding session even if NvEncodeAPICreateInstance
succeeded. Asking driver about the version would be the most certain way.
The receive bin should block buffers from reaching dtlsdec before
the dtls connection has started.
While there was code to block its sinkpads until receive_state
was different from BLOCK, nothing was ever setting it to BLOCK
in the first place. This commit corrects this by setting the
initial state to BLOCK, directly in the constructor.
In addition, now that blocking is effective, we want to only
block buffers and buffer lists, as that's what might trigger
errors, we want to still let events and queries go through,
not doing so causes immediate deadlocks when linking the
bin.
And free data with the correct free() function in the receive callback
by passing it to gst_buffer_new_wrapped_full() instead of
gst_buffer_new_wrapped().
alignment works like in mpegtsmux, joining several MpegTS packets into
one buffer. Default value of 0 joins as many as possible for each
incoming buffer, to optimise CPU usage.
