Martin Storsjö [Fri, 6 Mar 2015 10:27:14 +0000 (12:27 +0200)]
movenc: Keep writing zero-entry stts atoms as intended
a876585215 had the unintended side effect of returning AVERROR(ENOMEM)
when track->entry is zero, while the code intentionally wants to
continue in that case.
The mov muxer already supports picking up extradata that wasn't
present during the avformat_write_header call - we just need to
propagate it. Since the dash muxer uses delay_moov, we have time
up until the first segment is written to get extradata filled in.
Also update the codec description string when the extradata becomes
available.
avformat/adxdec: set avctx->channels in adx_read_header
It is used in adx_read_packet, which currently depends on the
decoder/parser setting this value between reading the file header and
demuxing the first packet.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
Martin Storsjö [Wed, 4 Mar 2015 07:52:33 +0000 (09:52 +0200)]
rtpdec_asf: Don't set RTP_FLAG_KEY
Nothing in the framework nor in the rest of the depacketizer actually
uses this flag - the chained demuxer sets the keyframe flag properly on
demuxed packets already.
Although the specification mandates this bit to zero, it may happen
that software tools incorrectly flip it to one, invalidating a possibly
valid stream.
Relax this restriction, by failing only when AV_EF_BITSTREAM is set.
This behaviour is similar to aac decoders in Firefox and Quicktime.
Similarly to what has been done for MOV, display XMP metadata only when
users explicitly require it.
The Extensible Metadata Platform tag can contain various kind of data
which are not strictly related to the video file, such as history of
edits and saves from the project file.
Vittorio Giovara [Fri, 27 Feb 2015 19:00:25 +0000 (19:00 +0000)]
aic: Fix decoding files with odd dimensions
Normally the aic decoder finds the proper slice combination (multiple of
some number less than 32) but in case of odd width, it resorts to the
default values, which were actually swapped.
The number of slices is modified to account for such odd width cases.
This commit broke playback of fragmented mp4 files with b-frames.
While investigating this, it turned out that the general framework
isn't ready for a PTS-based index yet. Revert this change until
a better thought out solution is in place.
Martin Storsjö [Wed, 25 Feb 2015 22:00:39 +0000 (00:00 +0200)]
rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate
Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.
This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)
All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.
For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.
This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).
Martin Storsjö [Thu, 26 Feb 2015 11:33:59 +0000 (13:33 +0200)]
rtpenc: Always do the default initialization regardless of codecs
This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).
Martin Storsjö [Wed, 25 Feb 2015 21:55:58 +0000 (23:55 +0200)]
rtpenc_aac: Fix sending fragmented frames
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>
Martin Storsjö [Tue, 24 Feb 2015 15:01:48 +0000 (17:01 +0200)]
rtpdec: Rename the free method to close
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.
Martin Storsjö [Tue, 24 Feb 2015 11:18:10 +0000 (13:18 +0200)]
rtpdec: Free depacketizers if the init function failed
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.
Martin Storsjö [Tue, 24 Feb 2015 11:07:57 +0000 (13:07 +0200)]
rtpdec: Don't free the payload context in the .free function
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.
Martin Storsjö [Mon, 23 Feb 2015 21:06:01 +0000 (23:06 +0200)]
rtpenc: Merge the h264 and hevc packetizers
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).
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets before each PLAY command to cause the router to
restart the port redirection in that case.
Move the existing code for sending the packets from the SETUP phase
to the PLAY phase.