- 06 3月, 2012 6 次提交
-
-
由 Justin Ruggles 提交于
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.
-
由 Anton Khirnov 提交于
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.
-
由 Anton Khirnov 提交于
Make packet buffer a parameter, don't hardcode it to be AVFormatContext.packet_buffer. Also move the function higher in the file, since it will be called from read_frame_internal().
-
由 Anton Khirnov 提交于
-
由 Fabian Greffrath 提交于
Signed-off-by: NDiego Biurrun <diego@biurrun.de>
-
由 Diego Biurrun 提交于
-
- 05 3月, 2012 5 次提交
-
-
由 Mans Rullgard 提交于
This splits ff_dsputil_init_mmx() into multiple functions, one for each MMX/SSE level, somewhat simplifying the nested conditions. Signed-off-by: NMans Rullgard <mans@mansr.com> Signed-off-by: NDiego Biurrun <diego@biurrun.de>
-
由 Anton Khirnov 提交于
Fixes listing encoders.
-
由 Anton Khirnov 提交于
-
由 Anton Khirnov 提交于
In most places where it's used, it's as a pointless write-only field. Only rv10 decoder actually reads from it, but it stores some internal version info in it. There is no reason for it to be in a public field.
-
由 Anton Khirnov 提交于
-
- 04 3月, 2012 29 次提交
-
-
由 Ronald S. Bultje 提交于
-
由 Kostya Shishkov 提交于
This allows user to select quantisation matrix from different profile, stamp frames with custom vendor string and change target bitrate.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
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.
-
由 Justin Ruggles 提交于
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.
-
由 Justin Ruggles 提交于
fixes writing of uninitialized packet data
-
由 Justin Ruggles 提交于
-
由 Aneesh Dogra 提交于
Signed-off-by: NRonald S. Bultje <rsbultje@gmail.com>
-
由 Ronald S. Bultje 提交于
Fixes pre-processing with latest versions of nasm.
-
由 Ronald S. Bultje 提交于
-
由 Ronald S. Bultje 提交于
-
由 Ronald S. Bultje 提交于
-
由 Ronald S. Bultje 提交于
-
由 Ronald S. Bultje 提交于
-
由 Ronald S. Bultje 提交于
-
由 Loren Merritt 提交于
Signed-off-by: NRonald S. Bultje <rsbultje@gmail.com>
-
由 Derek Buitenhuis 提交于
Signed-off-by: NDerek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: NJustin Ruggles <justin.ruggles@gmail.com>
-
由 Justin Ruggles 提交于
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. CC:libav-stable@libav.org
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
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.
-
由 Justin Ruggles 提交于
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. CC: libav-stable@libav.org
-
由 Justin Ruggles 提交于
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. CC:libav-stable@libav.org
-
由 Justin Ruggles 提交于
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. CC:libav-stable@libav.org
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
-
由 Justin Ruggles 提交于
Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.
-
由 Justin Ruggles 提交于
-
由 Justin Ruggles 提交于
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.
-
由 Justin Ruggles 提交于
Each packet has 18 sectors with 224/channels samples in each sector.
-