The rounding used in the PTS calculations in filter_frame() does
not actually match the number of samples output by the resampler.
This leads to off-by-1 errors in the timestamps indicating gaps and
underruns, even when the input timestamps are all contiguous.

Bug-Id: 753
Signed-off-by: Anton Khirnov <anton@khirnov.net>

It is a remnant of the old interrupt callback API.

Signed-off-by: Martin Storsjö <martin@martin.st>

Some RTMP commands need the most recent timestamp as their parameter, so
keep track of it. This must be the most recent one and not e.g. the max
received timestamp as it can decrease again through seeking.
Signed-off-by: Martin Storsjö <martin@martin.st>

For both audio and video.

For streams which contain DRC metadata, the FDK decoder is able to
control rendering of the decoded output. The rendering parameters
are detailed in fdk_aac_dec_options [].

The default behavior is left up to the decoder.
Signed-off-by: Martin Storsjö <martin@martin.st>

The FDK decoder is capable of producing mono and stereo downmix from
multichannel streams. These streams may contain metadata that control
the downmix process. The decoder requires an Ancillary Buffer in order to
correctly apply downmix in streams containing downmix Metadata. The
decoder does not have an API interface to inform of the presence of
Metadata in the stream, and therefore the Ancillary Buffer is always
allocated whenever a downmix is requested.

When downmixing multichannel streams, the decoder requires the output
buffer in aacDecoder_DecodeFrame call to be of fixed size in order to
hold the actual number of channels contained in the stream. For example,
for a 5.1ch to stereo downmix, the decoder requires that the output buffer
is allocated for 6 channels, regardless of the fact that the output is in
fact two channels.

Due to this requirement, the output buffer is allocated for the maximum
output buffer size in case a downmix is requested (and also during
decoder init). When a downmix is requested, the buffer used for output
during init will also be used for the entire duration the decoder is open.
Otherwise, the initial decoder output buffer is freed and the decoder
decodes straight into the output AVFrame.
Signed-off-by: Martin Storsjö <martin@martin.st>

In (non-live) streams with no metadata, the duration of a stream can
be retrieved by calling the RTMP function getStreamLength with the
playpath. The server will return a positive duration upon the request if
the duration is known, otherwise either no response or a duration of 0
will be returned.
Signed-off-by: Martin Storsjö <martin@martin.st>

Packets that contain a number as a result to a rtmp function call are
structured the same way (String, Number, Null, Number). This new method
also includes more bounds checks to better handle packets that are not
structured as expected.
Signed-off-by: Martin Storsjö <martin@martin.st>

The OptionDef arrays are terminated with a { NULL } element not NULL.

CC: libav-stable@libav.org
Bug-Id: CID 703769

Work as the other free()-like functions.

Bug-Id: CID 1087081
CC: libav-stable@libav.org

CC: libav-stable@libav.org
Bug-Id: CID 1224275

CC: libav-stable@libav.org
Bug-Id: CID 1005311

The element is always valid.

CC: libav-stable@libav.org
Bug-Id: CID 732276

CC: libav-stable@libav.org
Bug-Id: CID 733793

CC: libav-stable@libav.org
Bug-Id: CID 1238794

Signed-off-by: Martin Storsjö <martin@martin.st>

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Martin Storsjö <martin@martin.st>

Reported-by: Ruoyu <liangry@ucweb.com>

Reported-by: Ruoyu <liangry@ucweb.com>

Signed-off-by: Martin Storsjö <martin@martin.st>

These are assembled into extradata in the order vps/sps/pps/sei.
Signed-off-by: Martin Storsjö <martin@martin.st>

When decoding, this field holds the inverse of the framerate that can be
written in the headers for some codecs. Using a field called 'time_base'
for this is very misleading, as there are no timestamps associated with
it. Furthermore, this field is used for a very different purpose during
encoding.

Add a new field, called 'framerate', to replace the use of time_base for
decoding.

Decoding acceleration may work even if the codec level is higher than
the stated limit of the VDPAU driver. Or the problem may be considered
acceptable by the user. This flag allows skipping the codec level
capability checks and proceed with decoding.

Applications should obviously not set this flag by default, but only if
the user explicitly requested this behavior (and presumably knows how
to turn it back off if it fails).
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Currently, no flags are supported.
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Martin Storsjö <martin@martin.st>

These allow getting the absolute start timestamp of a fragment
without reading preceding timestamps. This fixes sync between
tracks if starting from fragments in different streams that don't
align exactly.

This also is a prerequisite for producing DASH content.
Signed-off-by: Martin Storsjö <martin@martin.st>

Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
  meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
  codec context is used for encoding or decoding (and has yet another
  different meaning for video), preventing generic handling of the codec
  context.

Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.