After sending a fragmented frame, len (s->buf_ptr - s->buf) isn't
zero, while s->num_frames is zero as intended. Using s->num_frames
makes it work as intended, and is less convoluted than keeping track
of (resetting) s->buf_ptr.
This avoids sending stray data after sending a fragmented aac packet.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Don't prefix them ffio_url, which is misleading, sounding too
much like the urlprotocol layer (like ffurl_*).
Signed-off-by: Martin Storsjö <martin@martin.st>
Many of these functions were named foo_free_context, and since
the functions no longer should free the context itself, only
allocated elements within it, the previous naming was slightly
misleading.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from how it is handled in codecs/demuxers/muxers
though (where the close function isn't called if the open function
failed), but since the number of depacketizers that have an .init
function is quite limited, this is easy to change.
The main point is that if the init function failed, we shouldn't
try to use that depacketizer at all - this makes sure that the
parse function doesn't need to check for the things that were
initialized in the init function.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes it more consistent with depacketizers that don't have any
.free function at all, where the payload context is freed by the
surrounding framework. Always free the context in the surrounding
framework, having the individual depacketizers only free any data
they've specifically allocated themselves.
This is similar to how this works for demuxer/muxers/codecs - a
component shouldn't free the priv_data that the framework has
allocated for it.
Signed-off-by: Martin Storsjö <martin@martin.st>
They share a great deal of common structure; only a few minor
bits in the headers differ.
This also fixes an off-by-one in sending of the last fragment
of large HEVC nals (where it previously sent len+2 bytes, even
if it should have been len+RTP_HEVC_HEADERS_SIZE aka len+3).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows getting rid of quite a bit of boilerplate in depacketizers.
The default value (initializing need_parsing to 0, aka
AVSTREAM_PARSE_NONE) is the same as it is initialized to by default
in AVStream.
Signed-off-by: Martin Storsjö <martin@martin.st>
If src_len is too small for nal_size, we already print a warning
above, and the next step is to check the while loop condition
anyway, so this one serves no purpose.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, errors were only logged but the code kept on trying,
and never actually returning the error as a return value.
Signed-off-by: Martin Storsjö <martin@martin.st>
Including libavcodec/get_bits.h is superfluous for AV_RB16 - nothing
in this file uses any actual bitstream reader.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows the output to be used with stream copy, which discards
packet from the start until the first keyframe.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes builds with vc1_parser enabled without vc1_decoder. All
the vc1_decoder object files were included in the vc1_parser line
in libavcodec/Makefile before, but architecture specific object files
for vc1_decoder were not.
Signed-off-by: Martin Storsjö <martin@martin.st>
The 1.3 release branch of OpenH264 (as well as the master branch)
have been updated so that GCC no longer warns about this variable
as being unused.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes sending chunked packets (packets larger than the output
chunk size, which often can be e.g. 4096 bytes) with a timestamp delta
(or absolute timstamp, if it's a timestamp step backwards, or the
first packet of the stream) larger than 0xffffffff.
The RTMP spec explicitly says (in section 5.3.1.3.) that packets of
type 3 (continuation packets) should include this field, if the
previous non-continuation packet had it included.
The receiving code handles these packets correctly.
Pointed out by Cheolho Park.
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>
Compared to existing, common opensource H264 encoders, this can be
useful since it has got a different license (BSD instead of GPL).
Performance- and qualitywise it is comparable to x264 in ultrafast
mode.
Hooking it up as an encoder in libavcodec also simplifies comparing
it against other common encoders.
This requires OpenH264 1.3 or newer. Since the OpenH264 API and ABI
changes frequently, only releases are supported.
To take advantage of the OpenH264 patent offer, the OpenH264 library
must not be redistributed, but downloaded at runtime at the end-user's
system.
Signed-off-by: Martin Storsjö <martin@martin.st>
This comment can be traced back to the initial commit from 2001,
and it seemed to be misleading/incorect already back then. (It
was used for normal, non-raw file formats already then.)
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows using libraries that are detected via pkg-config with
msvc. (The libraries themselves may have to be built with MSVC
though.)
Signed-off-by: Martin Storsjö <martin@martin.st>
When we don't adjust the Period start time, we don't need to
parse the earliest_presentation_time from the sidx boxes either.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was only necessary to get playback to start with dash.js 1.2.0,
it has been fixed in the git version.
The previous behaviour was incorrect - the Period's start time
is irrespective of the actual first timestamp of the contents
within the period. The Period start time only says when, within the
global timeline, this particular piece should start to be played
back.
Signed-off-by: Martin Storsjö <martin@martin.st>
This should be more correct. This also should give more sensible
switching between video streams with different amount of b-frame
delay.
The current dash.js release (1.2.0) fails to start playback of
such files from the start (if the start pts is > 0), but this has
been fixed in the current git version of dash.js.
Also enable the use of edit lists, so that streams in many cases
start at pts=0.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use the more generic approach with the delay_moov flag, instead of
having a update mechanism specific to this one single atom.
Signed-off-by: Martin Storsjö <martin@martin.st>
This delays writing the moov until the first fragment is written,
or can be flushed by the caller explicitly when wanted. If the first
sample in all streams is available at this point, we can write
a proper editlist at this point, allowing streams to start at
something else than dts=0. For AC3 and DNXHD, a packet is
needed in order to write the moov header properly.
This isn't added to the normal behaviour for empty_moov, since
the behaviour that ftyp+moov is written during avformat_write_header
would be changed. Callers that split the output stream into header+segments
(either by flushing manually, with the custom_frag flag set, or by
just differentiating between data written during avformat_write_header
and the rest) will need to be adjusted to take this option into use.
For handling streams that start at something else than dts=0, an
alternative would be to use different kinds of heuristics for
guessing the start dts (using AVCodecContext delay or has_b_frames
together with the frame rate), but this is not reliable and doesn't
necessarily work well with stream copy, and wouldn't work for getting
the right initialization data for AC3 or DNXHD either.
Signed-off-by: Martin Storsjö <martin@martin.st>
If fragments == 0 it means we haven't written any moov atom yet.
If the empty_moov flag is set, we already have written an empty moov
atom at startup. Thus, the check for empty_moov is redundant.
This is in preparation for allowing writing the moov atom later,
even when using the empty moov flag.
Signed-off-by: Martin Storsjö <martin@martin.st>
When writing an explicit time, reset the cur_time variable to this
value as well. This avoids writing excessive time attributes for each
segment in the timeline, as long as the segments are continuous.
Signed-off-by: Martin Storsjö <martin@martin.st>
The MoveFileExA is available in the headers regardless which API
subset is targeted, but it is missing in the Windows Phone link
libraries. When targeting Windows Store apps, the function is
available both in the headers and in the link libraries, and thus
there is no indication for the build system that this function
should be avoided - such an indication is only given by the
Windows App Certification Kit, which forbids using the MoveFileExA
function.
Therefore check the WINAPI_FAMILY defines instead, to figure out
which API subset is targeted.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the mpegts muxer now can handle being called with a NULL
AVIOContext, we don't need to try to allocate one before calling
write_trailer.
Signed-off-by: Martin Storsjö <martin@martin.st>
If opening and closing dynamic buffers as AVIOContext, we may
not have any AVIOContext available when wanting to close and
deallocate the muxer. Allow calling write_trailer despite this.
Signed-off-by: Martin Storsjö <martin@martin.st>
The pts and the corresponding duration is written in sidx
atoms, thus make sure these match up correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since this structurally is quite different from normal RTP
(multiple streams are muxed into one single mpegts stream,
which is packetized into one single RTP session), it is kept
as a separate muxer.
Since this structurally also behaves differently than normal
RTP, all of the other muxers that do chained RTP muxing
(rtsp, sap, mp4) would need to be updated similarly to handle
this - in particular, creating one single rtp_mpegts muxer
for the whole presentation instead of one rtp muxer per stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Instead explicitly jump to the default case in the cases where
it is wanted, and avoid fallthrough between different codecs,
which could easily introduce bugs if people editing the code
aren't careful.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids trying to do sliced encoding, even if a slice/packet
size is requested (via the -ps option or the rtp_payload_size
field), since the encoder currently doesn't support it (or at least
our decoder can't decode it, even if the h261_encode_gob_header
function is hooked up to be called from the slicing part in
mpegvideo_enc.c).
Signed-off-by: Martin Storsjö <martin@martin.st>
If we throw away the buffered incomplete frame, make sure to also
throw away the buffered bits of an incomplete byte at the same
time.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
In particular, when packetizing mpegts into rtp, the input packet
timestamp may come from more than one stream, which could cause
multiple packets be written with the same timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is the same adjustment that the mp4 muxer does to the start
timestamp of fragments, since the timestamp of a sample in an mp4
file is implicit from the sum of earlier sample durations.
This avoids gaps in the timeline (which can stop dash.js from
playing it back), and makes sure the timestamp on the segmenter
level matches what the mp4 muxer actually writes into the segments.
This is only an issue if the AVPacket duration of the last
packet of a segment doesn't point to the actual start timestamp
of the next packet (the first in the next segment).
Signed-off-by: Martin Storsjö <martin@martin.st>
Write a new start time if the duration of the previous segment
didn't match the start of the next one. Check that segments
actually are continuous before writing a repeat count.
This makes sure timestamps deduced from the timeline actually
match the real start timestamp as written in filenames (if
using a template containing $Time$).
Signed-off-by: Martin Storsjö <martin@martin.st>
Since 3cec81f4d4, a zero-length metadata value would try to
allocate 2*0 bytes, where av_malloc() returns NULL.
Always add one to the allocated length, to allow space for
a null terminator in the zero-length case.
Incidentally, this fixes fate-alac on RVCT 4.0, where a compiler
bug seems to mess up the mov muxer to the point that it writes
the wrong sort of metadata. Previously this bug was undetected,
but since 3cec81f4d4 such mov files started returning
AVERROR(ENOMEM) in the mov demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
For the last_duration field, it's mostly theoretical, but the
total_duration field more probably may need to actually be 64 bit.
Bug-Id: CID 1254944
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>
Only the upper 2 bits of the first byte are known to be
a fixed value.
The lower bits in the first byte of a RTP packet could be set
if the input is from another RTP packetizers than libavformat's,
but for RTCP packets, they would also be set when sending RTCP RR
packets, triggering false warnings about incorrect input format
to the protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
The RTP muxer enables the actual codepaths within sdp.c,
which depend on hevc.o since e5cfc8fd.
This fixes builds with --disable-everything --enable-muxer=rtp.
Signed-off-by: Martin Storsjö <martin@martin.st>
Martin Storsjö