FEC may only be used when PLC is enabled on the audio decoder,
as it relies on empty buffers to generate audio from the next
buffer. Hooking to the gap events doesn't work as the audio
decoder does not like more buffers output than it sends.
The length of data to generate using FEC from the next packet
is determined by rounding the gap duration to nearest. This
ensures that duration imprecision does not cause quantization
to 2.5 milliseconds less than available. Doing so causes the
Opus API to fail decoding. Such duration imprecision is common
in live cases.
The buffer to consider when determining the length of audio
to be decoded is the previous buffer when using FEC, and the
new buffer otherwise. In the FEC case, this means we determine
the amount of audio from the previous buffer, whether it was
missing or not (and get the data either from this buffer, or
the current one if the previous one was missing).
rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples
https://bugzilla.gnome.org/show_bug.cgi?id=759432
The result of the two expressions will be promoted to guint64 anyway,
perform all the arithmetic in 64 bits to avoid potential overflows.
CID 1338690, CID 1338691
We always require the channel-mapping-field. If it's 0 we require nothing
else, otherwise we need channels, stream-count and coupled count to be
available.
oggdemux is outputting the meta now, and only outputs if it should really
apply to the current buffer. Previously we would skip N samples also if we
started the decoder in the middle of the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
It is doing the wrong thing because of the Opus pre-skip: while the timestamps
are shifted by the pre-skip, the granule positions are not shifted.
oggmux is doing the right thing here already.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
The first frame has lookahead less samples, the last frame might have some
padding or we might have to encode another frame of silence to get all our
input into the encoded data.
This is because of a) the lookahead at the beginning of the encoding, which
shifts all data by that amount of samples and b) the padding needed to fill
the very last frame completely.
Ideally we would use LPC to calculate something better than silence for the
padding to make the encoding as smooth as possible.
With this we get exactly the same amount of samples again in an
opusenc ! opusdec pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
It is faster than doing a query that propagates downstream and
should be enough
Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
Previously, PLC frames always had a length of 120ms, which caused audio
quality degradation and synchronization errors. Fix this by calculating an
appropriate length for the PLC frame.
The length must be a multiple of 2.5ms. Calculate a multiple of 2.5ms that
is nearest to the current PLC length. Any leftover PLC length that didn't
make it into this frame is accumulated for the next PLC frame.
https://bugzilla.gnome.org/show_bug.cgi?id=725167
This way we let opusdec do the resampling if needed and don't carry
around buffers with a too high sample rate if not required.
While Opus always uses 48kHz internally, this information from the
header specifies which frequencies are safe to drop.
The max latency parameter is "the maximum time an element
synchronizing to the clock is allowed to wait for receiving all
data for the current running time" (docs/design/part-latency.txt).
https://bugzilla.gnome.org/show_bug.cgi?id=744338
