| Commit message (Collapse) | Author | Age | Files | Lines |
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If no streams were indicated in the FLV header, do not automatically
allocate by default a video and an audio stream. Instead, in the case
that the header did not indicate the presence of any data, allocate no
stream until data actually arrives for one type.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Some applications really mean to send negative pts.
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No point in sending the message multiple time.
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Messages should start with a capital letter and possibly end with a "."
if they are statements.
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The Omnia A/XE encoder writes the explicit extra data incorrectly
and wrongly disables parametric stereo. Truncating the extra data
by setting the size to 2 works around this. The AAC extra data
parser will then only parse the correct parts.
Bug-id: 599
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Certain streaming servers do not preserve the order of the fields.
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It is used internally.
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This is needed in order for the FLV demuxer not to detect a codec change when
using the "flv_metadata" option.
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The callers of this function can't report errors sanely. If this
one malloc fails, don't write the extradata byte, make sure we
try to malloc it the next time we're called instead, and make sure
we still consume the input data byte.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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This header byte is only present when actually reading a VP6 frame,
not when reading the codec type field in the metadata. This
potential bug has been present since 5b54a90c.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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New streams may be created at any time, e.g. on codec change.
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Signed-off-by: Martin Storsjö <martin@martin.st>
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We match streams by codec id now.
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This avoids creating new AVStreams for them when switching between
different variants of them, since we can handle changes between
different sample rates of nellymoser within the same stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
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In certain conditions video or audio frames might appear way later in the
stream.
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Codec change midstream gets mapped to a separate stream.
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Adobe specifies onTextData as the standard message to use to deliver
text information.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The index validation identifies these indexes as broken.
Signed-off-by: Martin Storsjö <martin@martin.st>
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By validating the index entries while reading, we don't need to
seek at startup to validate the entries. If the error in the
index entries is not pointing to (our definition of) the start
of packets, and there is an index entry pointing at some of the
first packets after the metadata, the invalid index can be discarded
almost immediately.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Also split a long line.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The demuxers don't use it in any way.
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Previously, we've only passed the key string on to the recursive
amf_parse_object for the mixedarray type, not for 'object'. By
passing the key string on, the recursive amf_parse_object can
store the amf objects as metadata.
This kind of data was seen in data from XSplit Broadcaster, received
over RTMP via Wowza. This patch allows reading this metadata.
Signed-off-by: Martin Storsjö <martin@martin.st>
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These can't be used uninitialized in practice, but the
compiler doesn't realize it.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Compared to just overwriting the old extradata, this has the
advantage of letting the decoder know exactly when the
extradata changed (otherwise it is changed immediately when the
new extradata packet is demuxed, even if there's old queued packets
awaiting to be decoded). This makes it easier for decoders to
actually react to the change, so they won't have to inspect
the extradata for each packet to see if it might have changed.
This works when sequentially playing a file with sample rate
changes, but if seeking past a new extradata packet in the
file, it obviously doesn't work properly. That case doesn't
work in flash player either, so it's probably ok not to handle
it.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
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It's supposed to be called only from (de)muxers.
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Specifically, ff_mpeg4audio_sample_rates, ff_mpeg4audio_get_config and
ff_copy_pce_data
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Manual replacements are done in this commit.
In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
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This avoids writing these entries doubly if transcoding from
flv to flv, since the muxer blindly writes any and all metadata
keys set, in addition to the fixed fields that the muxer
always writes.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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On allocation, the array length is multiplied by sizeof(int64_t),
this prevents the multiplication from overflowing.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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