- 27 10月, 2011 7 次提交
-
-
由 Justin Ruggles 提交于
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore the encoder based on the decoder is also incorrect. There is no good reason to keep the encoder.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
It is already checked by avcodec_open2().
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
the sample size.
-
由 Justin Ruggles 提交于
-
- 26 10月, 2011 21 次提交
-
-
由 Justin Ruggles 提交于
This is already done in avcodec_decode_audio3()
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
Also return AVERROR_PATCHWELCOME.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
They only operate on stereo content, so the extra param is not necessary and also allows for simplifying the code.
-
由 Justin Ruggles 提交于
Now they only do stereo interleaving.
-
由 Justin Ruggles 提交于
This should also fix decoding of mono 24-bit.
-
由 Justin Ruggles 提交于
It is identical for 16-bit and 24-bit, so there is no need to have duplicate code.
-
由 Justin Ruggles 提交于
The bits are not wasted, they are additional low bits that are added to the 16-bit decompressed samples to increase the output sample depth.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
Also rearranges some functions for easier cleanup on failure.
-
由 Justin Ruggles 提交于
reduces memory usage when the stream has fewer than MAX_CHANNELS
-
由 Justin Ruggles 提交于
check frame header channel count against header/container channel count.
-
由 Justin Ruggles 提交于
It is already done when using it to set sample_fmt.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Janne Grunau 提交于
Otherwise the delay expressed in has_b_frames increases with every avcodec_close/avcodec_open. Fixes fate-ea-dct with more than 1 thread.
-
由 Daniel Kang 提交于
Add whitespace. Signed-off-by: NRonald S. Bultje <rsbultje@gmail.com>
-
- 25 10月, 2011 12 次提交
-
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
It is broken because an AVCodecContext can be opened/closed multiple times, and sample_rate is getting divided by 2 each time that happens. This removes the only use of lowres for audio.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
using a floating-point calculation is not necessary.
-
由 Justin Ruggles 提交于
Note that this will not work in most cases with avconv and avplay due to the AVCODEC_MAX_AUDIO_FRAME_SIZE limit, but it will decode correctly if given a large enough output buffer.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
TTA does not support float at all, and format 2 is encrypted TTA.
-
由 Justin Ruggles 提交于
-