* qatar/master: (31 commits)
cdxl demux: do not create packets with uninitialized data at EOF.
Replace computations of remaining bits with calls to get_bits_left().
amrnb/amrwb: Remove get_bits usage.
cosmetics: reindent
avformat: do not require a pixel/sample format if there is no decoder
avformat: do not fill-in audio packet duration in compute_pkt_fields()
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
dca_parser: parse the sample rate and frame durations
libspeexdec: do not set AVCodecContext.frame_size
libopencore-amr: do not set AVCodecContext.frame_size
alsdec: do not set AVCodecContext.frame_size
siff: do not set AVCodecContext.frame_size
amr demuxer: do not set AVCodecContext.frame_size.
aiffdec: do not set AVCodecContext.frame_size
mov: do not set AVCodecContext.frame_size
ape: do not set AVCodecContext.frame_size.
rdt: remove workaround for infinite loop with aac
avformat: do not require frame_size in avformat_find_stream_info() for CELT
avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
avformat: do not require frame_size in avformat_find_stream_info() for AAC
...
Justin Ruggles [Mon, 27 Feb 2012 20:54:41 +0000 (15:54 -0500)]
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
Justin Ruggles [Mon, 27 Feb 2012 03:55:52 +0000 (22:55 -0500)]
aiffdec: do not set AVCodecContext.frame_size
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
Justin Ruggles [Mon, 27 Feb 2012 07:34:14 +0000 (02:34 -0500)]
riffenc: use av_get_audio_frame_duration()
For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
Justin Ruggles [Thu, 19 Jan 2012 23:36:40 +0000 (18:36 -0500)]
avcodec: add av_get_exact_bits_per_sample() function
This only returns bits per sample when it is exactly correct. That is, the
codec contains only raw samples with no frame headers or padding. This applies
to basically all PCM codecs and a small subset of ADPCM codecs.
Revert "wmaenc: check final frame size against output packet size"
This condition cannot happen, if it can it is a bug that MUST be fixed.
And i very happily volunteer to fix it if someone reports a case to
me that fails.
Anton Khirnov [Sun, 4 Mar 2012 14:49:26 +0000 (15:49 +0100)]
lavf: deobfuscate read_frame_internal().
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
Nicolas George [Wed, 15 Feb 2012 18:15:45 +0000 (19:15 +0100)]
timefilter: internally compute feedback factors.
The feedback factors for the timefilter are directly computed from
the expected period. This commit changes the init function to accept
the period itself and compute the feedback factors internally,
rather than having all client code duplicate the formulas.
This commit also actually fixes the formulas: the current code had
sqrt(2*o), but the correct formula, both theoretically and according
to experimental testing, is sqrt(2)*o.
Furthermore, it adds an exponential to feedback factors larger than
1 with large periods.
bit_depth_template: use av_clip_uint8 over crop_tab.
This fixes some global out of array reads and wrong cliping.
No speed difference meassurable under clang on i5
also all important code paths on all important platforms should
use SIMD.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
cmdutils: use new avcodec_is_decoder/encoder() functions.
lavc: make codec_is_decoder/encoder() public.
lavc: deprecate AVCodecContext.sub_id.
libcdio: add a forgotten AVClass to the private context.
swscale: remove "cpu flags" from -sws_flags description.
proresenc: give user a possibility to alter some encoding parameters
vorbisenc: add output buffer overwrite protection
libopencore-amrnbenc: fix end-of-stream handling
ra144enc: fix end-of-stream handling
nellymoserenc: zero any leftover packet bytes
nellymoserenc: use proper MDCT overlap delay
qpeg: Use bytestream2 functions to prevent buffer overreads.
swscale: make %rep unconditional.
vp8: convert simple loopfilter x86 assembly to use named arguments.
vp8: convert idct x86 assembly to use named arguments.
vp8: convert mc x86 assembly to use named arguments.
vp8: convert loopfilter x86 assembly to use cpuflags().
vp8: convert idct/mc x86 assembly to use cpuflags().
swscale: remove now unnecessary hack.
x86inc: don't "bake" stack_offset in named arguments.
...
Justin Ruggles [Sun, 4 Mar 2012 05:25:45 +0000 (00:25 -0500)]
libopencore-amrnbenc: fix end-of-stream handling
Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush
the encoder at the end of encoding. This is needed in order to have all input
samples decoded.
Justin Ruggles [Tue, 28 Feb 2012 06:02:28 +0000 (01:02 -0500)]
ra144enc: fix end-of-stream handling
Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush
the encoder at the end of encoding. This is needed in order to have all input
samples decoded.
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Justin Ruggles [Fri, 2 Mar 2012 22:11:25 +0000 (17:11 -0500)]
wmaenc: fix m/s stereo encoding for the first frame
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.
ffm options should also set discard automatically.
commit 13f6917ca91dfdc0fd785235b2dae891a9604859 handles discards automatically,
but the ffm discard options are not fully parsed. Causing the input streams not
to be used, so no stream towards the ffserver after the initial probing.
Signed-off-by: Rick van der Zwet <info@rickvanderzwet.nl> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Justin Ruggles [Fri, 2 Mar 2012 21:42:21 +0000 (16:42 -0500)]
wmaenc: check final frame size against output packet size
Currently we have an assert() that prevents the frame from being too large,
but it is more user-friendly to give an error message instead of aborting on
assert(). This condition is quite unlikely due to the minimum bit rate check
in encode_init(), but it is still worth having.
Justin Ruggles [Fri, 2 Mar 2012 21:33:33 +0000 (16:33 -0500)]
wmaenc: require a large enough output buffer to prevent overwrites
The maximum theoretical frame size is around 17000 bytes. Although in
practice it will generally be much smaller, we require a larger buffer
just to be safe.
Justin Ruggles [Fri, 2 Mar 2012 21:27:57 +0000 (16:27 -0500)]
wmaenc: limit allowed sample rate to 48kHz
ff_wma_init() allows up to 50kHz, but this generates an exponent band
size table that requires 65 bands. The code assumes 25 bands in many
places, and using sample rates higher than 48kHz will lead to buffer
overwrites.
Justin Ruggles [Fri, 2 Mar 2012 21:10:00 +0000 (16:10 -0500)]
wmaenc: limit block_align to MAX_CODED_SUPERFRAME_SIZE
This is near the theoretical limit for wma frame size and is the most that
our decoder can handle. Allowing higher bit rates will just end up padding
each frame with empty bytes.
Fixes invalid writes for avconv when using very high bit rates.
Justin Ruggles [Wed, 11 Jan 2012 15:22:47 +0000 (10:22 -0500)]
ea: fix audio pts
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.