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Replace ffmpeg references with more accurate libav* references.
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@ -36,7 +36,7 @@
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* a little more compression by exploiting the fact that adjacent pixels
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* tend to be similar.
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*
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* Note that this decoder could use ffmpeg's optimized VLC facilities
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* Note that this decoder could use libavcodec's optimized VLC facilities
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* rather than naive, tree-based Huffman decoding. However, there are 256
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* Huffman tables. Plus, the VLC bit coding order is right -> left instead
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* or left -> right, so all of the bits would have to be reversed. Further,
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@ -241,7 +241,7 @@ static av_cold int encode_init(AVCodecContext* avc_context)
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header, comment, and tables.
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Each one is prefixed with a 16bit size, then they
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are concatenated together into ffmpeg's extradata.
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are concatenated together into libavcodec's extradata.
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*/
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offset = 0;
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@ -27,7 +27,7 @@
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#include "avcodec.h"
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/*
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* Adapted to ffmpeg by Francois Revol <revol@free.fr>
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* Adapted to libavcodec by Francois Revol <revol@free.fr>
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* (removed 68k REG stuff, changed types, added some statics and consts,
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* libavcodec api, context stuff, interlaced stereo out).
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*/
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@ -1854,7 +1854,7 @@ static int decode_user_data(MpegEncContext *s, GetBitContext *gb){
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}
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}
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/* ffmpeg detection */
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/* libavcodec detection */
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e=sscanf(buf, "FFmpe%*[^b]b%d", &build)+3;
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if(e!=4)
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e=sscanf(buf, "FFmpeg v%d.%d.%d / libavcodec build: %d", &ver, &ver2, &ver3, &build);
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@ -30,8 +30,8 @@
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* Note that this decoder reads big endian RGB555 pixel values from the
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* bytestream, arranges them in the host's endian order, and outputs
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* them to the final rendered map in the same host endian order. This is
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* intended behavior as the ffmpeg documentation states that RGB555 pixels
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* shall be stored in native CPU endianness.
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* intended behavior as the libavcodec documentation states that RGB555
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* pixels shall be stored in native CPU endianness.
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*/
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#include <stdio.h>
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@ -19,7 +19,7 @@
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*/
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/* The *no_round* functions have been added by James A. Morrison, 2003,2004.
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The vis code from libmpeg2 was adapted for ffmpeg by James A. Morrison.
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The vis code from libmpeg2 was adapted for libavcodec by James A. Morrison.
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*/
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#include "config.h"
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@ -2,9 +2,9 @@
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* Duck Truemotion v1 Decoding Tables
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*
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* Data in this file was originally part of VpVision from On2 which is
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* distributed under the GNU GPL. It is redistributed with ffmpeg under the
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* GNU LGPL using the common understanding that data tables necessary for
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* decoding algorithms are not necessarily licensable.
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* distributed under the GNU GPL. It is redistributed with libavcodec under
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* the GNU LGPL using the common understanding that data tables necessary
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* for decoding algorithms are not necessarily copyrightable.
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*
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* This file is part of Libav.
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*
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@ -362,7 +362,7 @@ static int asf_read_stream_properties(AVFormatContext *s, int64_t size)
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/* Extract palette from extradata if bpp <= 8 */
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/* This code assumes that extradata contains only palette */
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/* This is true for all paletted codecs implemented in ffmpeg */
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/* This is true for all paletted codecs implemented in libavcodec */
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if (st->codec->extradata_size && (st->codec->bits_per_coded_sample <= 8)) {
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int av_unused i;
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#if HAVE_BIGENDIAN
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@ -35,7 +35,7 @@
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/* if we don't know the size in advance */
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#define AU_UNKNOWN_SIZE ((uint32_t)(~0))
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/* The ffmpeg codecs we support, and the IDs they have in the file */
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/* The libavcodec codecs we support, and the IDs they have in the file */
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static const AVCodecTag codec_au_tags[] = {
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{ CODEC_ID_PCM_MULAW, 1 },
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{ CODEC_ID_PCM_S8, 2 },
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@ -1,5 +1,5 @@
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/*
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* Various simple utilities for ffmpeg system
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* various simple utilities for libavformat
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* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
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*
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* This file is part of Libav.
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@ -96,7 +96,7 @@ static const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t)
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/*
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* There's a couple of assumptions being made here:
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* 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
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* We can pass them upwards when ffmpeg will be ready to deal with them.
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* We can pass them upwards when libavcodec will be ready to deal with them.
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* 2. We don't do software emphasis.
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* 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
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* are converted into 16bit linear ones.
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@ -26,7 +26,7 @@
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#define RSO_HEADER_SIZE 8
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/* The ffmpeg codecs we support, and the IDs they have in the file */
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/* The libavcodec codecs we support, and the IDs they have in the file */
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extern const AVCodecTag ff_codec_rso_tags[];
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#endif /* AVFORMAT_RSO_H */
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