| Commit message (Collapse) | Author | Age | Files | Lines |
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6050 b64f7644-9d1e-0410-96f1-a4d463321fa5
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replace it with libavcodec ac3 decoder
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6049 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6042 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Remove some patches that have been applied upstream.
Add support for AVDownmixInfo side data (i.e. mix levels).
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6039 b64f7644-9d1e-0410-96f1-a4d463321fa5
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-1 is not a good value as a flag for invalid timestamps.
There are cases where small negative timestamps are useful.
So this eliminates a potential ambiguity.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6001 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6000 b64f7644-9d1e-0410-96f1-a4d463321fa5
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These fields correspond to the actual audio data in the AVFrame, and should always be set (otherwise, we must submit a bug report to libav).
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5994 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Patch originally by the Multicoreware Inc team, followed by improvements and fixes by Micheal Wootton from AMD Inc,
OpenCL:
This patch implements Bicubic Scaling in OpenCL.
Note that HandBrake currently uses Lanczos so the performance difference appears to be much more significant. We may offer an option of BiCubic in software later.
Bicubic scaling may appear a bit sharper than the equivalent Lanczos encode and may increase file size a bit. Quality may be better or worse depending on the scaling and content and personal preference towards sharpness.
When comparing performance with a custom HandBrake build that runs Software Bicubic to OpenCL Bicubic, performance increase is about 5~7% on average on a modern GPU.
Hardware Decode via DXVA:
We also have optional DXVA decoding which may come in useful for slower/lower end systems that have a capable GPU.
This is only available on input sources that use the libav decode path.
Most GPU hardware for decoding is designed for playback, so if you are running on a high end CPU, it will bottleneck the encode process.
Requires OpenCL 1.1 or later supporting GPU.
Front end changes and testing framework are not included in this patch. This will be resolved later.
Patch will be revised further before the UI is implemented.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5792 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5318 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5074 b64f7644-9d1e-0410-96f1-a4d463321fa5
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- encavcodecaudio: use libavresample directly (instead of via the hb_audio_resample wrapper), and add support for planar output
- hb_audio_resample: add support for planar input
- hb_audio_remap: add support for planar input
- deca52: output planar float samples (no re-interleaving)
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5073 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5029 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5028 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Some functions were called once per block, where once per frame was enough.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5027 b64f7644-9d1e-0410-96f1-a4d463321fa5
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- support for additional sample formats
- support for variable channel layouts
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5023 b64f7644-9d1e-0410-96f1-a4d463321fa5
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This matches liba52's built-in downmix code.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4966 b64f7644-9d1e-0410-96f1-a4d463321fa5
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hb_audio_resample_init().
Unlike hb_audio_remap, hb_audio_resample will never be used in situations where we don't have a mixdown.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4923 b64f7644-9d1e-0410-96f1-a4d463321fa5
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dedicated functions.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4922 b64f7644-9d1e-0410-96f1-a4d463321fa5
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an error when it fails, so we know which decoder actually called the function.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4904 b64f7644-9d1e-0410-96f1-a4d463321fa5
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liba52.
Add support for center & surround mix levels to hb_audio_resample.
This allows us to support upmixing all audio sources. For sources that have at least 2 front and one back or side channel(s), allow upmixing to 5.1:
3.0/3.1 (2 front and 1 back channels)
4.0/4.1 (3 front and 1 back channels)
4.0/4.1 (2 front and 2 side channels)
5.0 (3 front and 2 side channels)
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4885 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Resolves several deprecated api's
Eliminates several libav patches
Eliminates our builtin downmix in favour of avresample
Eliminate HB_INPUT_CH_LAYOUT_* and replace with AV_CH_LAYOUT_*
Resolves 6.x and 7.0 input channel layout issues HB had
Adds downmix support to declpcm. We never had it!
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4825 b64f7644-9d1e-0410-96f1-a4d463321fa5
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AV_CH_LAYOUT_* works for us, and as its usage becomes more widespread in libhb, translating between the two formats gets increasingly tedious.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4754 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4737 b64f7644-9d1e-0410-96f1-a4d463321fa5
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necessary and didn't work in all decoders for Dolby downmixing.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4714 b64f7644-9d1e-0410-96f1-a4d463321fa5
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doing it wrong in some cases.
This means Dolby Pro Logic II is now only available for sources with at least L, R, C, Ls, Rs channels (as well as any additional channels), due to some libdca limitations. This will eventually go away when we use our own downmixing for all sources.
Further cleanup to follow.
Also cleans up usage of audio->config.in.flags (was audio->config.flags.*).
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4705 b64f7644-9d1e-0410-96f1-a4d463321fa5
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sync needs the precise duration of a frame of audio and has
no way to compute it for passthru audio. So add a duration
member to hb_buffer_t and set it in the audio decoders.
This should eventually supersede hb_buffer_t member "stop"
since it provides redundant information.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4618 b64f7644-9d1e-0410-96f1-a4d463321fa5
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This patch enhances the filter objects. The 2 key improvements are:
1. A filter can change the image dimensions as frames pass through it.
2. A filter can output more than one frame.
In addition, I have:
Moved cropping & scalling into a filter object
Added 90 degree rotation to the rotate filter
Moved subtitle burn-in rendering to a filter object.
Moved VFR/CFR handling into a framerate shaping filter object.
Removed render.c since all it's responsibilities got moved to filters.
Improves VOBSUB and SSA subtitle handling. Allows subtitle animations.
SSA karaoke support.
My apologies in advance if anything breaks ;)
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4546 b64f7644-9d1e-0410-96f1-a4d463321fa5
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adds aac and mp3 passthru for mp4 and mkv containers
adds dts and dtshd for mp4 container (mkv already had it)
Note: The only player known (to me) to support dts(hd) in mp4 is ff/avplay
In LinGui there is a new option to limit which passthru codecs
will be used by the "Auto Passthru" audio codec options. The CLI
already has this ability with "--audio_copy-mask" which is use
in conjunction with the "copy" audio codec option.
Also corrects some A/V sync issues when video frames are dropped due to
a gap detected in the audio.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4149 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Decoders set the channel map of their output in hb_audio_config_t.
Encoders use this information to remap while encoding.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4052 b64f7644-9d1e-0410-96f1-a4d463321fa5
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...from float [-32768...32767] to float [-1.0...1.0]
Using the range [-1.0..1.0] requires fewer translations of the range for our
various encoders and decoders. This also gets rid of a hacky
translation from float to int to float in decavcodec audio decoding.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@3908 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Uses ffmpeg's ac3 encoder.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@3570 b64f7644-9d1e-0410-96f1-a4d463321fa5
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enable mono mode in lame encoder when mixdown is mono.
use lame_encode_buffer_float instead of lame_encode_buffer_interleave. this
eliminates the clipping issue in lame without reducing the level of the input.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@3294 b64f7644-9d1e-0410-96f1-a4d463321fa5
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The lame codec has a more limited input range than the other codecs. So
tell the downmixers to stay strictly within the level range requested.
also changes some tabs into spaces in deca52 and decdca
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@3205 b64f7644-9d1e-0410-96f1-a4d463321fa5
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remains the same (default DRC) and >1 still boosts soft sounds.
For the moment, presets and interfaces still assume a default of 1, but the notion is to move to having 0 as the default, in the hope it will be louder/clearer.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@2119 b64f7644-9d1e-0410-96f1-a4d463321fa5
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crc) rather than looking at multiple frames. This should reduce the probability of mis-identifying random junk as AC-3 to less than 1 in 2^32.
- check the crc on every frame so we don't let corrupted data into the decoder.
- interpret the PTS as per the standard (it's the time of the first frame that starts in the packet, not the time of the first byte of the packet). Incorrect interpretation was resulting in an average 15ms timing error (worse case 31ms).
- do all the PTS calculations in doubles so we don't get round-off error that will desync the audio & video with 44.1KHz audio sources (these can't appear on DVDs but do show up in avi/mkv/... files).
- don't rely on the container to give us large enough frame fragments to validate the sync (some containers split audio frames into really small pieces). Instead use the 8K of unused space in the esconfig of the audio object as an accumulation buffer.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@2002 b64f7644-9d1e-0410-96f1-a4d463321fa5
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that was hardcoding the AC3 stream information in the dac3 atom based on the AppleTV startup movie stream.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1879 b64f7644-9d1e-0410-96f1-a4d463321fa5
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buffer.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1850 b64f7644-9d1e-0410-96f1-a4d463321fa5
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matching syncs.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1838 b64f7644-9d1e-0410-96f1-a4d463321fa5
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check for two AC3 packets in a row that agree on their values before accepting them as gospel.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1832 b64f7644-9d1e-0410-96f1-a4d463321fa5
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pieces so we don't get jerky output caused by out-of-order frames.
- add an explicit EOF for all streams, not just video.
- don't generate extra audio silence at the end of an encode (don't need it with explicit eof).
- get rid of 80ms initial delay in AAC encode & flush final four frames buffered in encoder.
- put mp4 'chap' atom on first track (usually video) rather than first audio track since we can now do video without audio (atom just needs to go on an enabled media track & video is always enabled).
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1581 b64f7644-9d1e-0410-96f1-a4d463321fa5
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- support video files handled by ffmpeg (avi, mkv, mp4, etc.)
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1480 b64f7644-9d1e-0410-96f1-a4d463321fa5
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covers trunk files, libhb, and test.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1418 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Each audio track contains it's own settings for codec, bitrate, samplerate, etc.
This allows for very complex combinations of audio tracks if desired.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1365 b64f7644-9d1e-0410-96f1-a4d463321fa5
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start and stop time which caused bad things to happen later on in sync.c. Move up the code that generates a pts for non-passthru frames so that it applies to both.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1340 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1307 b64f7644-9d1e-0410-96f1-a4d463321fa5
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for the same.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1267 b64f7644-9d1e-0410-96f1-a4d463321fa5
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value is greater than 1, rather than non-zero. This prevents calling the function when it's only going to return the default values anyway.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1124 b64f7644-9d1e-0410-96f1-a4d463321fa5
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types.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1117 b64f7644-9d1e-0410-96f1-a4d463321fa5
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interface, making softer sounds louder. Based on code from liba52's A52Decoder, used in Perian's preferences pane. Set with a job->dynamic_range_compression float, with valid values being 1.0 - 4.0.
Controlled from the CLI with --dynamic-range-compression or -D.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1108 b64f7644-9d1e-0410-96f1-a4d463321fa5
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sequence numbers to determine where in the stream the buffer came from.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@1011 b64f7644-9d1e-0410-96f1-a4d463321fa5
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