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.
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.
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.
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.
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.
Justin Ruggles [Fri, 13 Jan 2012 19:04:11 +0000 (14:04 -0500)]
vqf: set packet duration
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
Justin Ruggles [Fri, 24 Feb 2012 23:04:05 +0000 (18:04 -0500)]
vorbisdec: read the previous window flag for long windows
When reading sequentially, we are using the actual flag from the previous
frame, but when seeking we do not know what the previous window flag was, so
we need to read it from the bitstream.
Martin Storsjö [Fri, 2 Mar 2012 15:03:06 +0000 (17:03 +0200)]
g722: Fix the QMF scaling
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.
This makes the decoder output have double the magnitude
compared to before.
The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.
Anton Khirnov [Thu, 1 Mar 2012 20:24:16 +0000 (21:24 +0100)]
electronicarts: set timebase for tgv video.
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.