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>
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>
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.
Currently, the API takes an external AVCodecContext, which is used only
for extradata and logging. This change will allow to it to work without
an AVCodecContext in the following commits.
It is never an error if this method failed. If rt->live was
explicitly set to 0 (known to be a recorded file), print it
as a warning, otherwise print it as a debug message.
Based on a patch by Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
The application will destroy the underlying hardware handles when
get_format() gets called again. Also this ensures the
deinitialization takes place if the get_format callback returns an
error.
Regression from 1c80c9d7ef.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The tfdt atom shouldn't be needed in those cases, we already
write tfxd atoms for ismv anyway, which is roughly equivalent.
This avoids having to declare the iso6 brand for ismv files.
Signed-off-by: Martin Storsjö <martin@martin.st>
ISO/IEC 14496-12:2012/Cor 1:2013 is explicit about how this should be
handled. All zeros doesn't mean that the full file has got a zero
duration, only that the track samples described within the initial moov
have got zero duration. An all ones duration means an indeterminate
duration.
Keep writing a duration consisting of all ones for the ISM mode -
older windows media player versions won't play a file if this is
zero. (Newer windows media player versions play either version fine.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Similarly to the omit_tfhd_offset flag added in e7bf085b, this
avoids writing absolute byte positions to the file, making them
more easily streamable.
This is a new feature from 14496-12:2012, so application support
isn't necessarily too widespread yet (support for it in libav was
added in 20f95f21f in July 2014).
Signed-off-by: Martin Storsjö <martin@martin.st>
The custom IO flag actually never is set for muxers, only for
demuxers, so the check was pointless (unless a user intentionally
would set the flag to signal using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
If one track doesn't have any samples within a moof, no traf/trun
is written for it. When the omit_tfhd_offset flag is set, none
of the tfhd atoms have any base_data_offset set, and the implicit
offset (end of previous track fragment data, or start of the moof
for the first trun) is used.
Signed-off-by: Martin Storsjö <martin@martin.st>
