A flag "dash" is added, which enables the necessary flags for
creating DASH compatible fragments.
When this is enabled, one sidx atom is written for each track
before every moof atom.
Signed-off-by: Martin Storsjö <martin@martin.st>
By calling this after writing the moof the first time (for
calculating the moof size), we can avoid intermediate storage
of tfrf_offset in MOVTrack.
Signed-off-by: Martin Storsjö <martin@martin.st>
When writing fragmented streams with an empty initial moov,
we won't have any samples in any tracks when writing the
moov atom, thus trust that any tracks that are added actually
will be present.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is a generic solution that will not reqiore modifications when new options are added.
This also fixes problem with current implementation when qmin or qmax=-1.
Only 8 bits was sent and read back as 255.
Fixes#1275Fixes#1461
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
In http_open_cnx, the patch restores the AVDictionary if connection needs to be re-tried
because of a authentication/redirect status code.
Previously, if a 401/407/30x status code was encountered, http_open_cnx would restart at the redo label, but any options
used by the underlying protocol would be missing because they were removed by the first attempt.
Signed-off-by: Brandon Lees <brandon@n-hega.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is needed because Icecast since version 2.4.1 doesn't default
to audio/mpeg anymore. AVOption default not used here, since a later
check if -content_type is set is performed and would break.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This allows for proper error reporting. Only do
this for non-legacy requests as only Icecast >2.4.0
will reply with a proper status.
Libav seems to accept both, 100 and 200 status codes, but
let's stay close to spec.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
According to the DASH spec, Representation IDs should be unique
across all adaptation sets. Fixing that and updating the fate
reference file to reflect this change.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Using 100-continue ffmpeg will only send data if the server confirms it,
so if there is an error with auth or mounpoint, this allows that it is
properly reported to the user. Else ffmpeg sends data and just quits at
some point without an error message.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
use a default (audio/mpeg for historical reason) if none. Required since Icecast 2.4.1
Not using AVOption default because this breaks content-type warnings (needs to
detect if no type was set by the user)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows selecting if the demuxer should consider all streams to be
found after the first PMT and add further streams during decoding or if it rather
should scan all that are within the analyze-duration and other limits
Fixes Ticket3762
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids a potential crash if writing a fragmented psp mp4
(which probably is only a hypothetical scenario).
Signed-off-by: Martin Storsjö <martin@martin.st>
In matroska_read_seek(), |tracks| is assigned at the begining of the function.
However, functions like matroska_parse_cues() could reallocate the tracks so
that |tracks| can get invalidated.
This CL assigns |tracks| only before we use it so that it won't be invalidated.
BUG=427266
TEST=Test case in associated bug passes now.
Change-Id: I9c7065fe8f4311ca846076281df2282d190ed344
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously we wrote decoding timestamps here, while the specs
say it should be presentation timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
When using the new first_trun flag instead of checking the track id,
we don't need to have a special case for the separate_moof flag
any longer.
This simplifies the complicated codepath ever so slightly.
Signed-off-by: Martin Storsjö <martin@martin.st>
In this case, shift tracks to start from zero instead (potentially
stretching the first sample in tracks that start later than the
first one).
Some software does not support edit lists at all, the adobe flash
player seems to be one of these. This results in AV sync errors when
edit lists are used to adjust AV sync.
Some players, such as QuickTime, don't respect the duration for
audio packets, so if an audio track starts later than the video
track and the first audio sample gets a duration longer than the
actual amount of data in it, the result will be out of sync.
Based on patches by Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is the same logic as is invoked on AVFMT_TS_NEGATIVE,
but which can be enabled manually, or can be enabled
in muxers which only need it in certain conditions.
Also allow using the same mechanism to force streams to start
at 0.
Signed-off-by: Martin Storsjö <martin@martin.st>
The only parameters needed by the demuxers are the sample rate and sample
count, which can be trivially extracted manually, without resorting to
an avpriv function.
It will not be set unless the codec context is used as the encoding
context, which is discouraged. For MP2, av_get_audio_frame_duration()
will already set the frame size properly. For MP3, set the frame size
explicitly.
Martin Storsjö
Michael Niedermayer
Andrew Stone
Lukasz Marek
Brandon Lees
Mika Raento
Marvin Scholz
Mark Reid
Vittorio Giovara
Peter Ross
Carl Eugen Hoyos
Vignesh Venkatasubramanian
Xiaohan Wang
Anton Khirnov