The timeDataSize argument to aacDecoder_DecodeFrame() seems undocumented
and until 2016 04 (203e3f28fbebec7011342017fafc2a0bda0ce530) unused.
After that commit libfdk-aacdec interprets it as size in sample units
and memsets that on error.
FFmpeg as well as others (like GStreamer) did interpret it as size in
bytes.
Fixes: 1442/clusterfuzz-testcase-minimized-4540199973421056 (This requires recent libfdk to reproduce)
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously most of the error paths leaked.
Also add FF_CODEC_CAP_INIT_THREADSAFE while adding caps_internal;
this decoder wrapper doesn't have any static data that is initialized.
Signed-off-by: Martin Storsjö <martin@martin.st>
For ADTS streams, the output format (number of channels, frame size)
can change at any point (with the latest version of fdk-aac, the decoder
seems to change format after a handful of frames, not outputting the
right format immediately, for cases that worked fine with the earlier
version of the lib).
Previously, the decoder decoded straight into the output frame once the
number of channels and frame size was known. This obviously does not
work if the number of channels or frame size changes.
The alternative would be to allocate the AVFrame with the maximum number
of channels and frame size, and change them afterward decoding into it,
but that may cause confusion to users e.g. of the get_buffer callback.
This solution should be more robust.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
In the latest version of fdk-aac, the decoder can output up to 8
channels; take this into account when preallocating buffers that
need to fit the output from any packet.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The latest version added support for a new option for enabling
a signal level limiter, which adds some extra delay. In fdk-aac, this
is enabled by default, but disable it by default here since we'd rather
have zero-delay decoding.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids potential out of bounds writes, with potential future
versions of the library.
Bug-Id: CID 1254945
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The latest fdk-aac code drop (from android 5.0) changed the channel
layout enums (changing the value of existing enum constants), and
renamed the option for downmixing.
The failsafe comparison between ctype and FF_ARRAY_ELEMS(channel_counts)
can trigger warnings (-Wtautological-constant-out-of-range-compare)
when building with the old FDK AAC releases, where it can't be
out of range with the enum values used there.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
For implicit signaling cases (as possible for Spectral Band Replication
and Parametric Stereo Tools), the decoder must decode the first frame to
correctly identify the stream configuration (as called from
avformat_find_stream_info). The mechanism for this is built-in and only
requires adding CODEC_CAP_CHANNEL_CONF to the libfdk-aacdec AVCodec
struct.
Signed-off-by: Omer Osman <omer.osman@iis.fraunhofer.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
The default error concealment method if none is set via
aacDecoder_SetParam(AAC_CONCEAL_METHOD) is set in
CConcealment_InitCommonData within the fdk-aac library
and is set to Energy Interpolation. This method requires one frame
delay to the output. To reduce the default decoder output delay and
avoid missing the last frame in file based decoding, use Noise
Substitution as the default concealment method.
Signed-off-by: Omer Osman <omer.osman@iis.fraunhofer.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
This reduces the number of different licenses used within libav,
and is preferrable since it has less ambiguous wordings than
the BSD license with respect to the duties of the user of the code.
Fraunhofer have now indicated that they're allowed to contribute
code under this license as well.
Signed-off-by: Martin Storsjö <martin@martin.st>
This should make it possible for Fraunhofer to contribute to these
wrappers - they didn't want to contribute to code under LGPL2.1 with
the "or any later version" clause (which allowed using the code
under the LGPL3 license).
Signed-off-by: Martin Storsjö <martin@martin.st>
Michael Niedermayer
Diego Biurrun
Vittorio Giovara
Martin Storsjö
Omer Osman