To ensure the full range of values is still used, also adjust all uses of this function to loop from 0
instead of 1. This way only 60.00 is added and nothing lost.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fix hardcoded tables compililation caused by missing math constants
lavf: Make codec_tag arrays constant
twinvq: give massive struct a name.
lavf, lavu: version bumps and APIchanges for av_gettime() move
lavfi/audio: don't set cur_buf in ff_filter_samples().
lavfi/fifo: add audio version of the fifo filter.
fifo: fix parenthesis placement.
lavfi: rename vf_fifo.c -> fifo.c
lavc: remove stats_in from AVCodecContext options table.
Conflicts:
doc/APIchanges
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/audio.c
libavfilter/fifo.c
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This muxer supports CODEC_ID_SRT with the timestamps in the packet data
and CODEC_ID_TEXT with the timestamps in the packet fields.
Makes -scodec copy work from Matroska.
* qatar/master:
MS Screen 1 decoder
aacdec: Fix popping channel layouts.
av_gettime: support Win32 without gettimeofday()
Use av_gettime() in various places
Move av_gettime() to libavutil
dct-test: use emms_c() from libavutil instead of duplicating it
mov: fix operator precedence bug
mathematics.h: remove a couple of math defines
Remove unnecessary inclusions of [sys/]time.h
lavf: remove unnecessary inclusions of unistd.h
bfin: libswscale: add const where appropriate to fix warnings
bfin: libswscale: remove unnecessary #includes
udp: Properly check for invalid sockets
tcp: Check the return value from getsockopt
network: Use av_strerror for getting error messages
udp: Properly print error from getnameinfo
mmst: Use AVUNERROR() to convert error codes to the right range for strerror
network: Pass pointers of the right type to get/setsockopt/ioctlsocket on windows
rtmp: Reduce the number of idle posts sent by sleeping 50ms
Conflicts:
Changelog
configure
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dct-test.c
libavcodec/version.h
libavformat/riff.c
libavformat/udp.c
libavutil/Makefile
libswscale/bfin/yuv2rgb_bfin.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (24 commits)
flvdec: remove incomplete, disabled seeking code
mem: add support for _aligned_malloc() as found on Windows
lavc: Extend the documentation for avcodec_init_packet
flvdec: remove incomplete, disabled seeking code
http: replace atoll() with strtoll()
mpegts: remove unused/incomplete/broken seeking code
af_amix: allow float planar sample format as input
af_amix: use AVFloatDSPContext.vector_fmac_scalar()
float_dsp: add x86-optimized functions for vector_fmac_scalar()
float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
lavr: Add x86-optimized function for flt to s32 conversion
lavr: Add x86-optimized function for flt to s16 conversion
lavr: Add x86-optimized functions for s32 to flt conversion
lavr: Add x86-optimized functions for s32 to s16 conversion
lavr: Add x86-optimized functions for s16 to flt conversion
lavr: Add x86-optimized function for s16 to s32 conversion
rtpenc: Support packetizing iLBC
rtpdec: Add a depacketizer for iLBC
Implement the iLBC storage file format
mov: Support muxing/demuxing iLBC
...
Conflicts:
Changelog
configure
libavcodec/avcodec.h
libavcodec/dsputil.c
libavcodec/version.h
libavformat/movenc.c
libavformat/mpegts.c
libavformat/version.h
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also use ff_neterrno() instead of errno directly (which doesn't work
on windows), for getting the error code.
Signed-off-by: Martin Storsjö <martin@martin.st>
getnameinfo doesn't set errno on failure, it returns an error code,
which should be handled by gai_strerror instead of the normal
strerror.
Signed-off-by: Martin Storsjö <martin@martin.st>
Rtmpt is effectively half duplex - the server can't return any
data unless we send a request (to which the server responds). If
we don't have any data to send currently, and the server didn't
return any data either, wait a little before doing the next request.
This avoids busy looping with idle posts with empty replies, while
waiting for more data from the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new option 'rtmp_flush_interval' that allows specifying the
number of packets to write before sending it off as a HTTP request.
This is mostly relevant for RTMPT - for plain RTMP, it only controls
how often we check the socket for incoming packets, which shouldn't
affect the performance in any noticeable way.
Signed-off-by: Martin Storsjö <martin@martin.st>
This seems to be the correct mode to send, according to the
original RTSP RFC, and matches the method RECORD which is
sent later when starting to send data.
Darwin Streaming Server works fine with either of them.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs
lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs
Add Dolby/DPLII downmix support to libavresample
vorbisdec: replace div/mod in loop with a counter
fate: vorbis: add 5.1 surround test
rtpenc: Allow requesting H264 RTP packetization mode 0
configure: Sort the library listings in the help text alphabetically
dwt: remove variable-length arrays
RTMPT protocol support
http: Properly handle chunked transfer-encoding for replies to post data
http: Fail reading if the connection has gone away
amr: Mark an array const
amr: More space cleanup
rtpenc: Fix memory leaks in the muxer open function
Conflicts:
Changelog
configure
doc/APIchanges
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This requires all NAL units to fit within single RTP packets. It
doesn't change the actual packetization for packets that fit, but
errors out and gives a helpful hint if the NAL units would have to
be split, and signals the right packetization mode in the SDP.
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can happen if doing a new request using the same socket,
but the new request failed, which clears the urlcontext.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add spaces around operators, fix brace placement and whitespace to
match K&R style, vertically align code, remove redundant != 0 and
convert x == 0 into !x, drop useless braces.
Signed-off-by: Martin Storsjö <martin@martin.st>
Defining restrict results - for some compilers - in changing other
uses of the restrict keyword also, e.g. __declspec(restrict) gets
changed to __declspec(__restrict) on MSVC. This causes compilation
failures. Therefore, using a private namespace macro instead is
more reliable and robust.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
flacdec: read attached pictures.
lavf: don't segfault when a NULL filename is passed to avformat_open_input()
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This can easily happen when the caller is using a custom AVIOContext.
Behave as if the filename was an empty string in this case.
CC: libav-stable@libav.org
* qatar/master:
af_resample: fix format modifier in debug string for FF_API_SAMPLERATE64
segment: remove unnecessary <strings.h> include
fate: add snow hpel tests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
currently a overflow there should be impossible but future changes to
the code could easily introduce a bug that no longer limits the 2
values sufficiently so better protect it via av_assert.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Avoid C99 variable declarations within for statements.
rtmp: Read and handle incoming packets while writing data
doc: document THREAD_TYPE fate variable
rtpdec: Don't require frames to start with a Mode A packet
avconv: don't try to free threads that were not initialized.
Conflicts:
doc/fate.texi
ffplay.c
libavdevice/dv1394.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure all incoming packets are read and handled (and reacted
to) while sending an FLV stream over RTMP to a server. If there were
enough incoming data to fill the TCP buffers, this could potentially
make things block at unexpected places. For the upcoming RTMPT support,
we need to consume all incoming data before we can send the next
request.
Signed-off-by: Martin Storsjö <martin@martin.st>
While there is no reason for starting a frame with anything else
than a Mode A packet, some senders seem to consistently use Mode B
packets for everything. This fixes depacketization of such streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtmp: Add a new option 'rtmp_buffer', for setting the client buffer time
rtmp: Set the client buffer time to 3s instead of 0.26s
rtmp: Handle server bandwidth packets
rtmp: Display a verbose message when an unknown packet type is received
lavfi/audio: use av_samples_copy() instead of custom code.
configure: add all filters hardcoded into avconv to avconv_deps
avfiltergraph: remove a redundant call to avfilter_get_by_name().
lavfi: allow building without swscale.
build: Do not delete tests/vsynth2 directory, which is no longer created.
lavfi: replace AVFilterContext.input/output_count with nb_inputs/outputs
lavfi: make AVFilterPad opaque after two major bumps.
lavfi: add avfilter_pad_get_type() and avfilter_pad_get_name().
lavfi: make avfilter_get_video_buffer() private on next bump.
jack: update to new latency range API as the old one has been deprecated
rtmp: Tokenize the AMF connection parameters manually instead of using strtok_r
ppc: Rename H.264 optimization template file for consistency.
lavfi: add channelsplit audio filter.
golomb: check remaining bits during unary decoding in get_ur_golomb_jpegls()
sws: fix planar RGB input conversions for 9/10/16 bpp.
Conflicts:
Changelog
configure
doc/APIchanges
ffmpeg.c
libavcodec/golomb.h
libavcodec/v210dec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/asrc_anullsrc.c
libavfilter/audio.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/buffersrc.c
libavfilter/formats.c
libavfilter/version.h
libavfilter/vf_frei0r.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/video.h
libavfilter/vsrc_color.c
libavformat/rtmpproto.c
libswscale/input.c
tests/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This factorizes existing code into a new function gen_buffer_time(),
which generates the client buffer time message and sends it to the
server.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avfilter: Log an error if avfilter fails to configure a link.
avconv: support only native pthreads.
rtmp: Fix a possible access to invalid memory location when the playpath is too short.
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes an issue with a crazy data track starting with a large
negative timestamp.
It could as well be solved in all user apps, but this is looking
attractively simpler ...
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Do not send extension for flv files
rtmp: support connection parameters
doc: Add documentation for the newly added rtmp_* options
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow using connection parameters in order to append arbitrary
AMF data like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0" to the
Connect message. You can pass these parameters through the -rtmp_conn
option.
Signed-off-by: Martin Storsjö <martin@martin.st>
Apple softwares seem not to add a tref for the timecode (the next commit
fixes this issue), but at least FFmpeg does.
This can be used to generate a sample that demonstrates the feature:
./ffmpeg -f lavfi -i testsrc \
-f lavfi -i mptestsrc \
-f lavfi -i rgbtestsrc \
-map 0 -map 1 -map 2 \
-metadata:s:0 timecode=00:00:00:12 \
-metadata:s:2 timecode=01:02:12:20 \
-t 10 -y out.mov
./ffprobe out.mov
The timecode metadata being transmitted to the video streams, it can be
kept while transmuxed/transcoded.
* qatar/master:
h264: allow cropping to AVCodecContext.width/height
mov: set AVCodecContext.width/height for h264
iac: generate codec tables as they are supposed to be
indeo4: handle frame type 1 properly
lavu: change versioning script to include all av* prefixed symbols
Conflicts:
libavcodec/h264.c
libavutil/libavutil.v
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vorbis: Validate that the floor 1 X values contain no duplicates.
avprobe: Identify codec probe failures rather than calling them unsupported codecs.
avformat: Probe codecs at score 0 on buffer exhaustion conditions.
avformat: Factorize codec probing.
Indeo Audio decoder
imc: make IMDCT support stereo output
imc: move channel-specific data into separate context
lavfi: remove request/poll and drawing functions from public API on next bump
lavfi: make avfilter_insert_pad and pals private on next bump.
lavfi: make formats API private on next bump.
avplay: use buffersrc instead of custom input filter.
avtools: move buffer management code from avconv to cmdutils.
avconv: don't use InputStream in the buffer management code.
avconv: fix exiting when max frames is reached.
mpc8: fix maximum bands handling
aacdec: Turn PS off when switching to stereo and turn it to implicit when switching to mono.
Conflicts:
Changelog
cmdutils.h
ffmpeg.c
ffplay.c
ffprobe.c
libavcodec/avcodec.h
libavcodec/mpc8.c
libavcodec/v210dec.h
libavcodec/version.h
libavcodec/vorbisdec.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersrc.c
libavfilter/formats.c
libavfilter/src_movie.c
libavfilter/vf_aspect.c
libavfilter/vf_blackframe.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_cropdetect.c
libavfilter/vf_delogo.c
libavfilter/vf_drawbox.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_fifo.c
libavfilter/vf_format.c
libavfilter/vf_frei0r.c
libavfilter/vf_gradfun.c
libavfilter/vf_hflip.c
libavfilter/vf_hqdn3d.c
libavfilter/vf_libopencv.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_showinfo.c
libavfilter/vf_transpose.c
libavfilter/vf_unsharp.c
libavfilter/vf_yadif.c
libavfilter/vsrc_color.c
libavfilter/vsrc_testsrc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: Write chan atom for all audio tracks in mov mode movies.
mpegtsenc: use avio_open_dyn_buf(), zero pointers after freeing
doc/avconv: add some details about the transcoding process.
avidec: make scale and rate unsigned.
avconv: check output stream recording time before each frame returned from filters
avconv: split selecting input file out of transcode().
avconv: split checking for active outputs out of transcode().
avfiltergraph: make some functions static.
Conflicts:
ffmpeg.c
libavfilter/avfiltergraph.c
libavfilter/internal.h
libavformat/mpegtsenc.c
tests/ref/fate/acodec-alac
tests/ref/fate/acodec-pcm-s16be
tests/ref/fate/acodec-pcm-s24be
tests/ref/fate/acodec-pcm-s32be
tests/ref/fate/acodec-pcm-s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some demuxers set a timecode in the format or streams metadata. The
muxers now make use of this metadata instead of a duplicated private
option.
This makes possible transparent copy of the timecode when transmuxing
and transcoding.
-timecode option for MPEG1/2 codec is also renamed to -gop_timecode. The
global ffmpeg -timecode option will set it anyway so no option change
visible for the user.
* qatar/master:
librtmp: return AVERROR_UNKNOWN instead of -1.
librtmp: don't abuse a variable for two unrelated things.
librtmp: add rtmp_app and rtmp_playpath private options.
bmv: add stricter checks for invalid decoded length
avpacket: fix duplicating side data.
flv: support stream text data as onTextData
Conflicts:
libavcodec/bmv.c
libavformat/flvdec.c
libavformat/flvenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegtsenc: Support LATM packetization for AAC
adtsenc: Don't expose the muxer internals to the rest of lavf
mpegtsenc: use AVFormatContext for AAC packetization
mpegtsenc: use AVERROR() for return codes
Conflicts:
libavformat/adts.h
libavformat/mpegtsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds the avoption mpegts_flags and converts the existing
resend_headers option into a flag, keeping the old option as
fallback for now.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This isn't required any longer, when the mpegts muxer uses it
as a proper chained muxer.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This removes the dependency on adts.c internals, and simplifies
adding other packetization formats.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
http: Add the url_shutdown function for https, too
http: Simplify code by removing a local variable
http: Clear the old URLContext pointer when closed
tcp: Try enabling SO_REUSEADDR when listening
tcp: Check the return values from bind and accept
avisynth: Make sure the filename passed to avisynth is in the right code page
Conflicts:
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes issues with opening http urls that have authentication
or redirects, introduced in commit e999b641.
Signed-off-by: Martin Storsjö <martin@martin.st>
avisynth is a non-unicode application and cannot accept UTF-8
characters. Therefore, the input filename should be converted to
the correct code page that it expects.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
fate: Fix fate-ac3-fixed-encode for pre-ssse3 x86 machines
http: Pass the proper return code of net IO operations
http: Add 'post_data', a new option which sets custom HTTP post data
lavfi: amix: check active input count before calling request_samples
vp8: move block coeff arithcoder on stack.
mp3/ac3 probe: search for PES headers to prevent probing MPEG-PS as MP3.
Conflicts:
libavformat/ac3dec.c
libavformat/mp3dec.c
tests/fate/ac3.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f919cc7df6ab844bc12f89fe7bef4fb915a47725':
fate: fix acodec/vsynth tests for make 3.81
pcm_mpeg: fix number of consumed bytes to include the header.
avfilter: include required header file avfilter.h in video.h
x86: Avoid movs on BUTTERFLYPS when in AVX mode
x86: use new schema for ASM macros
fate: convert codec-regression.sh to makefile rules
fate: allow tests to specify unit size for psnr comparison
fate: teach videogen/rotozoom to output a single raw video stream
http: Add support for reusing the http socket for subsequent requests
http: Add support for using persistent connections
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Such files are currently not supported as the table is used at several points
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There is basic support for muxing chapter information into the
Apple Quicktime format already, but there are two errors which
prevent correct detection on the player side.
1) A special apple 'text' atom needs to be included inside the
gmhd atom.
2) The *different* 'text' atom inside the 'stsd' atom needs a
proper header.
With these changes, the chapters are now picked up by Apple
players and reported correctly by tools like mediainfo and mp4chaps.
v3 Update: The stub TextSampleEntry creation is moved to where the
chapter track is created so it's now specific to this track.
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Introduce ff_http_do_new_request(), a new function which sends a new
HTTP request, reusing the existing connection to the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new AVOption 'multiple_requests', which indicates if we want
to use persistent connections (ie. Connection: keep-alive).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avprobe: restore pseudo-INI old style format for compatibility.
avprobe: fix formatting.
log: make colored output more colorful.
rtsp: Check for dynamic payload handlers if no static payload mapping was found
Conflicts:
Changelog
doc/ffprobe.texi
ffprobe.c
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
opt: Add av_opt_set_bin()
avconv: Display the error returned by avformat_write_header
rtpenc_chain: Return an error code instead of just a plain pointer
rtpenc_chain: Free the URLContext on failure
rtpenc: Expose the ssrc as an avoption
avprobe: display the codec profile in show_stream()
avprobe: fix function prototype
cosmetics: Fix indentation
avprobe: changelog entry
avprobe: update documentation
avprobe: provide JSON output
avprobe: output proper INI format
avprobe: improve formatting
rtmp: fix url parsing
fate: document TARGET_EXEC and its usage
Conflicts:
doc/APIchanges
doc/fate.texi
doc/ffprobe.texi
ffprobe.c
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some systems abuse the static payload types 35 or 36 (which
according to IANA are unassigned) for H264.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an URLContext is passed in, its ownership is given to this
function, and is either owned by the returned AVFormatContext
on a successful return, or freed on failure.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
movenc: Don't write the 'wave' atom or its child 'enda' for lpcm audio.
imc: some cosmetics
rtmp: Pass the proper return code in rtmp_handshake
rtmp: Check return codes of net IO operations
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Return a proper error code instead of -1
rtmp: Check malloc calls
rtmp: Check ff_rtmp_packet_create calls
lavfi: add audio mix filter
flvdec: Make sure sample_rate is set to the updated value
tqi: Pass errors from the MB decoder
Conflicts:
Changelog
doc/filters.texi
libavcodec/eatqi.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
os_support: Define SHUT_RD, SHUT_WR and SHUT_RDWR on OS/2
http: Add support for reading http POST reply headers
http: Add http_shutdown() for ending writing of posts
tcp: Allow signalling end of reading/writing
avio: Add a function for signalling end of reading/writing
lavfi: fix comment, audio is supported now.
lavfi: fix incorrect comment.
lavfi: remove avfilter_null_* from public API on next bump.
lavfi: remove avfilter_default_* from public API on next bump.
lavfi: deprecate default config_props() callback and refactor avfilter_config_links()
avfiltergraph: smarter sample format selection.
avconv: rename transcode_audio/video to decode_audio/video.
asyncts: reset delta to 0 when it's not used.
x86: lavc: use %if HAVE_AVX guards around AVX functions in yasm code.
dwt: return errors from ff_slice_buffer_init()
Conflicts:
ffmpeg.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/version.h
libavfilter/vf_blackframe.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_format.c
libavfilter/vf_showinfo.c
libavfilter/video.c
libavfilter/video.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
