Martin Storsjö [Wed, 22 Oct 2014 09:40:46 +0000 (12:40 +0300)]
lavf: Use av_gettime_relative
The ones left using av_gettime are NTP timestamps (for RTCP,
which is specified to send the actual current realtime clock
in RTCP SR packets), and the NUT muxer timestamper, which is
documented as using wallclock time.
Whenever av_gettime() is used to measure relative period of time,
av_gettime_relative() is prefered as it guarantee monotonic time
on supported platforms.
Whenever av_gettime() is used to measure relative period of time,
av_gettime_relative() is prefered as it guarantee monotonic time
on supported platforms.
Martin Storsjö [Wed, 22 Oct 2014 09:14:37 +0000 (12:14 +0300)]
lavu: Add av_gettime_relative
Since av_gettime() is used in a number of places where actual
real time clock is required, the monotonic clock introduced in ebef9f5a5 would have consequences that are hard to handle. Instead
split it into a separate function that can be used in the cases
where only relative time is desired.
On platform where no monotonic clock is available, the difference
between the two av_gettime functions is not clear, and one could
mistakenly use the relative clock where an absolute one is
required. Therefore add an offset, to make it evident that the
time returned from av_gettime_relative never is actual current
real time, even though it is based on av_gettime.
Anton Khirnov [Wed, 8 Oct 2014 10:49:11 +0000 (10:49 +0000)]
libvpxenc: add static-thresh private option
Currently, this option is accessed through AVCodecContext.mb_threshold,
which originally controlled reusing MB data when transcoding mpeg to
mpeg. Since the libvpx meaning is completely different from the original
mpegvideo meaning, it is better to use a separate private option for
this.
The rounding used in the PTS calculations in filter_frame() does
not actually match the number of samples output by the resampler.
This leads to off-by-1 errors in the timestamps indicating gaps and
underruns, even when the input timestamps are all contiguous.
Uwe L. Korn [Fri, 17 Oct 2014 14:30:46 +0000 (16:30 +0200)]
rtmpproto: Track last received timestamp
Some RTMP commands need the most recent timestamp as their parameter, so
keep track of it. This must be the most recent one and not e.g. the max
received timestamp as it can decrease again through seeking.
Omer Osman [Tue, 14 Oct 2014 14:43:08 +0000 (16:43 +0200)]
libfdk-aacdec: Enable Dynamic Range Control Metadata Support
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 [].
Omer Osman [Tue, 14 Oct 2014 14:43:07 +0000 (16:43 +0200)]
libfdk-aacdec: Enable Decoder Downmix including Downmix Metadata Support
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.
Uwe L. Korn [Tue, 14 Oct 2014 15:16:21 +0000 (17:16 +0200)]
rtmpproto: Add getStreamLength call to query duration
In (non-live) streams with no metadata, the duration of a stream can
be retrieved by calling the RTMP function getStreamLength with the
playpath. The server will return a positive duration upon the request if
the duration is known, otherwise either no response or a duration of 0
will be returned.
Uwe L. Korn [Tue, 14 Oct 2014 15:16:20 +0000 (17:16 +0200)]
rtmpproto: Add function to read a number response
Packets that contain a number as a result to a rtmp function call are
structured the same way (String, Number, Null, Number). This new method
also includes more bounds checks to better handle packets that are not
structured as expected.
Anton Khirnov [Fri, 4 Apr 2014 10:47:44 +0000 (12:47 +0200)]
lavc: deprecate the use of AVCodecContext.time_base for decoding
When decoding, this field holds the inverse of the framerate that can be
written in the headers for some codecs. Using a field called 'time_base'
for this is very misleading, as there are no timestamps associated with
it. Furthermore, this field is used for a very different purpose during
encoding.
Add a new field, called 'framerate', to replace the use of time_base for
decoding.
vdpau: add AV_HWACCEL_FLAG_IGNORE_LEVEL to skip the codec level check
Decoding acceleration may work even if the codec level is higher than
the stated limit of the VDPAU driver. Or the problem may be considered
acceptable by the user. This flag allows skipping the codec level
capability checks and proceed with decoding.
Applications should obviously not set this flag by default, but only if
the user explicitly requested this behavior (and presumably knows how
to turn it back off if it fails).
Martin Storsjö [Mon, 6 Oct 2014 20:20:50 +0000 (23:20 +0300)]
movenc: Write tfdt atoms
These allow getting the absolute start timestamp of a fragment
without reading preceding timestamps. This fixes sync between
tracks if starting from fragments in different streams that don't
align exactly.
This also is a prerequisite for producing DASH content.
Anton Khirnov [Sat, 23 Aug 2014 12:40:50 +0000 (12:40 +0000)]
lavc: use a separate field for exporting audio encoder padding
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
codec context is used for encoding or decoding (and has yet another
different meaning for video), preventing generic handling of the codec
context.
Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
Mark McGough [Sun, 12 Oct 2014 10:24:07 +0000 (18:24 +0800)]
icecast: Do not use chunked post
Icecast uses HTTP 1.0 while Libav uses HTTP 1.1 and enables by
default chunked post.
Icecast actually forwards the HTTP chunk headers to the listener
as part of the media stream (without the chunk encoding HTTP headers)
causing the players to lose sync.
Disabling the option is enough to feed icecast properly.
Martin Storsjö [Sat, 11 Oct 2014 21:03:25 +0000 (00:03 +0300)]
movenc: Write edit lists for fragmented files as well, if necessary
This is necessary to get the right timestamp offset for content
that starts with dts != 0.
This currently only helps when writing fragmented files with a non-empty
moov atom. When writing an empty moov atom, we don't have any packets
yet, so we don't know the starting dts for the tracks.
projects / ffmpeg / log