Merge commit '9d58639e270f7612874681e0ca30fa461e2667b7'

* commit '9d58639e270f7612874681e0ca30fa461e2667b7': sws: Drop deprecated SWS_CPU_CAPS defines Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
sws: Drop deprecated SWS_CPU_CAPS defines
2015-09-05 15:58:52 +02:00 · 2015-08-28 10:55:26 +02:00
656 changed files with 4227 additions and 13112 deletions
--- a/4
+++ b/4
@@ -1,6 +1,6 @@
-See the Git history of the project (https://git.ffmpeg.org/ffmpeg) to
+See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
 get the names of people who have contributed to FFmpeg.

 To check the log, you can type the command "git log" in the FFmpeg
 source directory, or browse the online repository at
-https://git.ffmpeg.org/ffmpeg
+http://source.ffmpeg.org.
--- a/1879
+++ b/1879
--- a/13
+++ b/13
@@ -56,7 +56,7 @@ Communication
 website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
 Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Baptiste Coudurier, Lou Logan
+mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
 Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
@@ -166,7 +166,7 @@ Codecs:
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dss_sp.c                              Oleksij Rempel
+  dss_sp.c                              Oleksij Rempel, Michael Niedermayer
  dv.c                                  Roman Shaposhnik
  dvbsubdec.c                           Anshul Maheshwari
  dxa.c                                 Kostya Shishkov
@@ -419,7 +419,7 @@ Muxers/Demuxers:
  cdxl.c                                Paul B Mahol
  crc.c                                 Michael Niedermayer
  daud.c                                Reimar Doeffinger
-  dss.c                                 Oleksij Rempel
+  dss.c                                 Oleksij Rempel, Michael Niedermayer
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
@@ -455,7 +455,7 @@ Muxers/Demuxers:
  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
-  mov.c                                 Baptiste Coudurier
+  mov.c                                 Michael Niedermayer, Baptiste Coudurier
  movenc.c                              Baptiste Coudurier, Matthieu Bouron
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
@@ -558,17 +558,17 @@ Windows MSVC                            Matthew Oliver, Hendrik Leppkes
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
-OS/2                                    KO Myung-Hun
+x86                                     Michael Niedermayer


 Releases
 ========

-2.8                                     Michael Niedermayer
 2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 2.4                                     Michael Niedermayer
+2.2                                     Michael Niedermayer

 If you want to maintain an older release, please contact us

@@ -596,7 +596,6 @@ Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-                              DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
--- a/2
+++ b/2
@@ -1 +1 @@
-2.8.22
+2.7.git
--- a/15
+++ b/15
@@ -1,15 +0,0 @@
-
-              ┌────────────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 2.8 "Feynman" │
-              └────────────────────────────────────────┘
-
-   The FFmpeg Project proudly presents FFmpeg 2.8 "Feynman", about 3
-   months after the release of FFmpeg 2.7.
-
-   A complete Changelog is available at the root of the project, and the
-   complete Git history on http://source.ffmpeg.org.
-
-   We hope you will like this release as much as we enjoyed working on it, and
-   as usual, if you have any questions about it, or any FFmpeg related topic,
-   feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
-   on the mailing-lists.
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -60,9 +60,6 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
-#if HAVE_SETDLLDIRECTORY
-#include <windows.h>
-#endif

 static int init_report(const char *env);

@@ -109,15 +106,6 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
    }
 }

-void init_dynload(void)
-{
-#if HAVE_SETDLLDIRECTORY
-    /* Calling SetDllDirectory with the empty string (but not NULL) removes the
-     * current working directory from the DLL search path as a security pre-caution. */
-    SetDllDirectory("");
-#endif
-}
-
 static void (*program_exit)(int ret);

 void register_exit(void (*cb)(int ret))
@@ -575,12 +563,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
        struct SwsContext *sws = sws_alloc_context();
        int ret = av_opt_set(sws, opt, arg, 0);
        sws_freeContext(sws);
-        if (!strcmp(opt, "srcw") || !strcmp(opt, "srch") ||
-            !strcmp(opt, "dstw") || !strcmp(opt, "dsth") ||
-            !strcmp(opt, "src_format") || !strcmp(opt, "dst_format")) {
-            av_log(NULL, AV_LOG_ERROR, "Directly using swscale dimensions/format options is not supported, please use the -s or -pix_fmt options\n");
-            return AVERROR(EINVAL);
-        }
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
            return ret;
@@ -2075,7 +2057,7 @@ static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
    if (!fmt || !fmt->priv_class  || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
        return AVERROR(EINVAL);

-    printf("Auto-detected sources for %s:\n", fmt->name);
+    printf("Audo-detected sources for %s:\n", fmt->name);
    if (!fmt->get_device_list) {
        ret = AVERROR(ENOSYS);
        printf("Cannot list sources. Not implemented.\n");
@@ -2105,7 +2087,7 @@ static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
    if (!fmt || !fmt->priv_class  || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
        return AVERROR(EINVAL);

-    printf("Auto-detected sinks for %s:\n", fmt->name);
+    printf("Audo-detected sinks for %s:\n", fmt->name);
    if (!fmt->get_device_list) {
        ret = AVERROR(ENOSYS);
        printf("Cannot list sinks. Not implemented.\n");
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -61,11 +61,6 @@ void register_exit(void (*cb)(int ret));
 */
 void exit_program(int ret) av_noreturn;

-/**
- * Initialize dynamic library loading
- */
-void init_dynload(void);
-
 /**
 * Initialize the cmdutils option system, in particular
 * allocate the *_opts contexts.
@@ -450,13 +445,13 @@ int show_devices(void *optctx, const char *opt, const char *arg);

 #if CONFIG_AVDEVICE
 /**
- * Print a listing containing autodetected sinks of the output device.
+ * Print a listing containing audodetected sinks of the output device.
 * Device name with options may be passed as an argument to limit results.
 */
 int show_sinks(void *optctx, const char *opt, const char *arg);

 /**
- * Print a listing containing autodetected sources of the input device.
+ * Print a listing containing audodetected sources of the input device.
 * Device name with options may be passed as an argument to limit results.
 */
 int show_sources(void *optctx, const char *opt, const char *arg);
--- a/16
+++ b/16
@@ -443,7 +443,7 @@ die(){

 If you think configure made a mistake, make sure you are using the latest
 version from Git.  If the latest version fails, report the problem to the
-ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.libera.chat.
+ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net.
 EOF
    if disabled logging; then
        cat <<EOF
@@ -1797,7 +1797,6 @@ SYSTEM_FUNCS="
    sched_getaffinity
    SetConsoleTextAttribute
    SetConsoleCtrlHandler
-    SetDllDirectory
    setmode
    setrlimit
    Sleep
@@ -2375,7 +2374,7 @@ zmbv_encoder_select="zlib"

 # hardware accelerators
 crystalhd_deps="libcrystalhd_libcrystalhd_if_h"
-d3d11va_deps="d3d11_h dxva_h ID3D11VideoDecoder ID3D11VideoContext"
+d3d11va_deps="d3d11_h dxva_h ID3D11VideoDecoder"
 dxva2_deps="dxva2api_h DXVA2_ConfigPictureDecode"
 vaapi_deps="va_va_h"
 vda_deps="VideoDecodeAcceleration_VDADecoder_h pthreads CoreServices_CoreServices_h"
@@ -5103,7 +5102,6 @@ check_func_headers windows.h MapViewOfFile
 check_func_headers windows.h PeekNamedPipe
 check_func_headers windows.h SetConsoleTextAttribute
 check_func_headers windows.h SetConsoleCtrlHandler
-check_func_headers windows.h SetDllDirectory
 check_func_headers windows.h Sleep
 check_func_headers windows.h VirtualAlloc
 check_struct windows.h "CONDITION_VARIABLE" Ptr
@@ -5149,7 +5147,6 @@ check_struct "sys/time.h sys/resource.h" "struct rusage" ru_maxrss

 check_type "windows.h dxva.h" "DXVA_PicParams_HEVC" -DWINAPI_FAMILY=WINAPI_FAMILY_DESKTOP_APP -D_CRT_BUILD_DESKTOP_APP=0
 check_type "windows.h d3d11.h" "ID3D11VideoDecoder"
-check_type "windows.h d3d11.h" "ID3D11VideoContext"
 check_type "d3d9.h dxva2api.h" DXVA2_ConfigPictureDecode -D_WIN32_WINNT=0x0602

 check_type "va/va.h" "VAPictureParameterBufferHEVC"
@@ -5242,7 +5239,7 @@ enabled libgsm            && { for gsm_hdr in "gsm.h" "gsm/gsm.h"; do
                                   check_lib "${gsm_hdr}" gsm_create -lgsm && break;
                               done || die "ERROR: libgsm not found"; }
 enabled libilbc           && require libilbc ilbc.h WebRtcIlbcfix_InitDecode -lilbc
-enabled libkvazaar        && require_pkg_config "kvazaar < 0.7.0" kvazaar.h kvz_api_get
+enabled libkvazaar        && require_pkg_config kvazaar kvazaar.h kvz_api_get
 enabled libmfx            && require_pkg_config libmfx "mfx/mfxvideo.h" MFXInit
 enabled libmodplug        && require_pkg_config libmodplug libmodplug/modplug.h ModPlug_Load
 enabled libmp3lame        && require "libmp3lame >= 3.98.3" lame/lame.h lame_set_VBR_quality -lmp3lame
@@ -5264,7 +5261,7 @@ enabled libshine          && require_pkg_config shine shine/layer3.h shine_encod
 enabled libsmbclient      && { use_pkg_config smbclient libsmbclient.h smbc_init ||
                               require smbclient libsmbclient.h smbc_init -lsmbclient; }
 enabled libsnappy         && require snappy snappy-c.h snappy_compress -lsnappy
-enabled libsoxr           && require libsoxr soxr.h soxr_create -lsoxr && LIBSOXR="-lsoxr"
+enabled libsoxr           && require libsoxr soxr.h soxr_create -lsoxr
 enabled libssh            && require_pkg_config libssh libssh/sftp.h sftp_init
 enabled libspeex          && require_pkg_config speex speex/speex.h speex_decoder_init -lspeex
 enabled libstagefright_h264 && require_cpp libstagefright_h264 "binder/ProcessState.h media/stagefright/MetaData.h
@@ -5737,7 +5734,6 @@ __declspec($_restrict) void* foo(int);
 EOF
    fi
    check_func strtoll || add_cflags -Dstrtoll=_strtoi64
-    check_func strtoull || add_cflags -Dstrtoull=_strtoui64
 fi

 for pfx in "" host_; do
@@ -6144,7 +6140,7 @@ cat > $TMPH <<EOF
 #define FFMPEG_CONFIG_H
 #define FFMPEG_CONFIGURATION "$(c_escape $FFMPEG_CONFIGURATION)"
 #define FFMPEG_LICENSE "$(c_escape $license)"
-#define CONFIG_THIS_YEAR 2023
+#define CONFIG_THIS_YEAR 2015
 #define FFMPEG_DATADIR "$(eval c_escape $datadir)"
 #define AVCONV_DATADIR "$(eval c_escape $datadir)"
 #define CC_IDENT "$(c_escape ${cc_ident:-Unknown compiler})"
@@ -6268,4 +6264,4 @@ pkgconfig_generate libavfilter   "FFmpeg audio/video filtering library" "$LIBAVF
 pkgconfig_generate libpostproc   "FFmpeg postprocessing library"        "$LIBPOSTPROC_VERSION"   ""
 pkgconfig_generate libavresample "Libav audio resampling library"       "$LIBAVRESAMPLE_VERSION" "$LIBM"
 pkgconfig_generate libswscale    "FFmpeg image rescaling library"       "$LIBSWSCALE_VERSION"    "$LIBM"
-pkgconfig_generate libswresample "FFmpeg audio resampling library"      "$LIBSWRESAMPLE_VERSION" "$LIBM $LIBSOXR"
+pkgconfig_generate libswresample "FFmpeg audio resampling library"      "$LIBSWRESAMPLE_VERSION" "$LIBM"
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,29 +15,27 @@ libavutil:     2014-08-09

 API changes, most recent first:

-------- 8< --------- FFmpeg 2.8 was cut here -------- 8< ---------
-
-2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h
+2015-xx-xx - lavc 56.58.100 - vaapi.h
  Deprecate old VA-API context (vaapi_context) fields that were only
  set and used by libavcodec. They are all managed internally now.

-2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h
+2015-xx-xx - lavu 54.31.100 - pixfmt.h
  Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that
  indicates the nature of the underlying storage: a VA surface. This
  yields the same value as AV_PIX_FMT_VAAPI_VLD.
  Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO,
  AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD.

-2015-08-02 - lavu 54.30.100 / 54.17.0
-  9ed59f1 / 7a7df34c -  Add av_blowfish_alloc().
-  a130ec9 / ae365453 -  Add av_rc4_alloc().
-  9ca1997 / 5d8bea3b -  Add av_xtea_alloc().
-  3cf08e9 / d9e8b47e -  Add av_des_alloc().
+2015-xx-xx - lavu 54.30.0
+  xxxxxxx -  Add av_blowfish_alloc().
+  xxxxxxx -  Add av_rc4_alloc().
+  xxxxxxx -  Add av_xtea_alloc().
+  xxxxxxx -  Add av_des_alloc().

-2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h
-  94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
-  444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
-  29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
+2015-xx-xx - lavc 56.35.0 - avcodec.h
+  xxxxxxxxx - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
+  xxxxxxxxx - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
+  xxxxxxxxx - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
              to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE.

 2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h
@@ -232,7 +230,7 @@ API changes, most recent first:
  Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
  dictionary types being set as options.

-2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
+2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
  Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
  the user when events happen in the file/stream.

@@ -249,7 +247,7 @@ API changes, most recent first:
 2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
  Add avio_feof() and deprecate url_feof().

-2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
+2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
  sws_getContext is not going to be removed in the future.

 2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         = 2.8.22
+PROJECT_NUMBER         =

 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
--- a/doc/authors.texi
+++ b/doc/authors.texi
@@ -3,9 +3,9 @@
 The FFmpeg developers.

 For details about the authorship, see the Git history of the project
-(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
@command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{https://git.ffmpeg.org/ffmpeg}.
+online repository at @url{http://source.ffmpeg.org}.

 Maintainers for the specific components are listed in the file
@file{MAINTAINERS} in the source code tree.
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -104,7 +104,7 @@ All subsequent file-related directives apply to that file.

@item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
-to 1 if it was -1.
+to 1 if it was to its default -1.

 To make FFmpeg recognize the format automatically, this directive must
 appears exactly as is (no extra space or byte-order-mark) on the very first
@@ -192,9 +192,7 @@ component.

 If set to 0, any file name is accepted.

-The default is 1.
-
-1 is equivalent to 1 if the format was automatically
+The default is -1, it is equivalent to 1 if the format was automatically
 probed and 0 otherwise.

@item auto_convert
@@ -283,24 +281,6 @@ used to end the output video at the length of the shortest input file,
 which in this case is @file{input.mp4} as the GIF in this example loops
 infinitely.

-@section hls
-
-HLS demuxer
-
-It accepts the following options:
-
-@table @option
-@item live_start_index
-segment index to start live streams at (negative values are from the end).
-
-@item allowed_extensions
-',' separated list of file extensions that hls is allowed to access.
-
-@item max_reload
-Maximum number of times a insufficient list is attempted to be reloaded.
-Default value is 1000.
-@end table
-
@section image2

 Image file demuxer.
@@ -436,23 +416,6 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize

-@section mov/mp4/3gp/Quicktme
-
-Quicktime / MP4 demuxer.
-
-This demuxer accepts the following options:
-@table @option
-@item enable_drefs
-Enable loading of external tracks, disabled by default.
-Enabling this can theoretically leak information in some use cases.
-
-@item use_absolute_path
-Allows loading of external tracks via absolute paths, disabled by default.
-Enabling this poses a security risk. It should only be enabled if the source
-is known to be non malicious.
-
-@end table
-
@section mpegts

 MPEG-2 transport stream demuxer.
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -400,35 +400,6 @@ finding a new maintainer and also don't forget updating the @file{MAINTAINERS} f

 We think our rules are not too hard. If you have comments, contact us.

-@section Code of conduct
-
-Be friendly and respectful towards others and third parties.
-Treat others the way you yourself want to be treated.
-
-Be considerate. Not everyone shares the same viewpoint and priorities as you do.
-Different opinions and interpretations help the project.
-Looking at issues from a different perspective assists development.
-
-Do not assume malice for things that can be attributed to incompetence. Even if
-it is malice, it's rarely good to start with that as initial assumption.
-
-Stay friendly even if someone acts contrarily. Everyone has a bad day
-once in a while.
-If you yourself have a bad day or are angry then try to take a break and reply
-once you are calm and without anger if you have to.
-
-Try to help other team members and cooperate if you can.
-
-The goal of software development is to create technical excellence, not for any
-individual to be better and "win" against the others. Large software projects
-are only possible and successful through teamwork.
-
-If someone struggles do not put them down. Give them a helping hand
-instead and point them in the right direction.
-
-Finally, keep in mind the immortal words of Bill and Ted,
-"Be excellent to each other."
-
@anchor{Submitting patches}
@section Submitting patches

--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -1353,7 +1353,7 @@ can be selected with @code{-pred 1}.
@table @option
@item format
 Can be set to either @code{j2k} or @code{jp2} (the default) that
-makes it possible to store non-rgb pix_fmts.
+allows to store non-rgb pix_fmts.

@end table

--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@} ... @{[@var{output_file_options}] @file{output_url}@} ...
+ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...

@chapter Description
@c man begin DESCRIPTION
@@ -24,10 +24,10 @@ rates and resize video on the fly with a high quality polyphase filter.
@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of output "files", which are
-specified by a plain output url. Anything found on the command line which
-cannot be interpreted as an option is considered to be an output url.
+specified by a plain output filename. Anything found on the command line which
+cannot be interpreted as an option is considered to be an output filename.

-Each input or output url can, in principle, contain any number of streams of
+Each input or output file can, in principle, contain any number of streams of
 different types (video/audio/subtitle/attachment/data). The allowed number and/or
 types of streams may be limited by the container format. Selecting which
 streams from which inputs will go into which output is either done automatically
@@ -243,8 +243,8 @@ Force input or output file format. The format is normally auto detected for inpu
 files and guessed from the file extension for output files, so this option is not
 needed in most cases.

-@item -i @var{url} (@emph{input})
-input file url
+@item -i @var{filename} (@emph{input})
+input file name

@item -y (@emph{global})
 Overwrite output files without asking.
@@ -277,7 +277,7 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
 When used as an input option (before @code{-i}), limit the @var{duration} of
 data read from the input file.

-When used as an output option (before an output url), stop writing the
+When used as an output option (before an output filename), stop writing the
 output after its duration reaches @var{duration}.

@var{duration} must be a time duration specification,
@@ -304,7 +304,7 @@ extra segment between the seek point and @var{position} will be decoded and
 discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
 will be preserved.

-When used as an output option (before an output url), decodes but discards
+When used as an output option (before an output filename), decodes but discards
 input until the timestamps reach @var{position}.

@var{position} must be a time duration specification,
@@ -1094,7 +1094,7 @@ may be reassigned to a different value.
 For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for
 an output mpegts file:
@example
-ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
+ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
@end example

@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
@@ -1205,9 +1205,9 @@ The option is intended for cases where features are needed that cannot be
 specified to @command{ffserver} but can be to @command{ffmpeg}.

@item -sdp_file @var{file} (@emph{global})
-Print sdp information for an output stream to @var{file}.
+Print sdp information to @var{file}.
 This allows dumping sdp information when at least one output isn't an
-rtp stream. (Requires at least one of the output formats to be rtp).
+rtp stream.

@item -discard (@emph{input})
 Allows discarding specific streams or frames of streams at the demuxer.
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffplay [@var{options}] [@file{input_url}]
+ffplay [@var{options}] [@file{input_file}]

@chapter Description
@c man begin DESCRIPTION
@@ -106,8 +106,8 @@ the input audio.
 Use the option "-filters" to show all the available filters (including
 sources and sinks).

-@item -i @var{input_url}
-Read @var{input_url}.
+@item -i @var{input_file}
+Read @var{input_file}.
@end table

@section Advanced options
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffprobe [@var{options}] [@file{input_url}]
+ffprobe [@var{options}] [@file{input_file}]

@chapter Description
@c man begin DESCRIPTION
@@ -24,8 +24,8 @@ For example it can be used to check the format of the container used
 by a multimedia stream and the format and type of each media stream
 contained in it.

-If a url is specified in input, ffprobe will try to open and
-probe the url content. If the url cannot be opened or recognized as
+If a filename is specified in input, ffprobe will try to open and
+probe the file content. If the file cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.

 ffprobe may be employed both as a standalone application or in
@@ -332,8 +332,8 @@ with name "PIXEL_FORMAT".
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.

-@item -i @var{input_url}
-Read @var{input_url}.
+@item -i @var{input_file}
+Read @var{input_file}.

@end table
@c man end
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1032,7 +1032,7 @@ Number_of_samples
 For example full key look like this @code{lavfi.astats.1.DC_offset} or
 this @code{lavfi.astats.Overall.Peak_count}.

-For description what each key means read below.
+For description what each key means read bellow.

@item reset
 Set number of frame after which stats are going to be recalculated.
@@ -2199,7 +2199,7 @@ Amount of milliseconds the signal has to rise above the threshold before gain
 reduction starts. Default is 20. Range is between 0.01 and 2000.

@item release
-Amount of milliseconds the signal has to fall below the threshold before
+Amount of milliseconds the signal has to fall bellow the threshold before
 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.

@item makeup
@@ -4118,7 +4118,7 @@ It accepts the following options:
@item threshold1
@item threshold2
@item threshold3
-Limit the maximum change for each plane, default is 65535.
+Allows to limit the maximum change for each plane, default is 65535.
 If 0, plane will remain unchanged.
@end table

@@ -4321,7 +4321,7 @@ It accepts the following options:
@item threshold1
@item threshold2
@item threshold3
-Limit the maximum change for each plane, default is 65535.
+Allows to limit the maximum change for each plane, default is 65535.
 If 0, plane will remain unchanged.

@item coordinates
@@ -4967,7 +4967,7 @@ within the parameter list.
@item
 Show the text at the center of the video frame:
@example
-drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"
+drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2"
@end example

@item
@@ -5206,7 +5206,7 @@ It accepts the following options:
@item threshold1
@item threshold2
@item threshold3
-Limit the maximum change for each plane, default is 65535.
+Allows to limit the maximum change for each plane, default is 65535.
 If 0, plane will remain unchanged.

@item coordinates
@@ -6094,7 +6094,7 @@ It accepts the following parameters:
@item filter_name
 The name of the frei0r effect to load. If the environment variable
@env{FREI0R_PATH} is defined, the frei0r effect is searched for in each of the
-directories specified by the colon-separated list in @env{FREI0R_PATH}.
+directories specified by the colon-separated list in @env{FREIOR_PATH}.
 Otherwise, the standard frei0r paths are searched, in this order:
@file{HOME/.frei0r-1/lib/}, @file{/usr/local/lib/frei0r-1/},
@file{/usr/lib/frei0r-1/}.
@@ -6900,7 +6900,7 @@ It accepts the following options:
@item threshold1
@item threshold2
@item threshold3
-Limit the maximum change for each plane, default is 65535.
+Allows to limit the maximum change for each plane, default is 65535.
 If 0, plane will remain unchanged.
@end table

@@ -9206,7 +9206,7 @@ uses the reference video instead of the main input as basis.

@itemize
@item
-Scale a subtitle stream (b) to match the main video (a) in size before overlaying
+Scale a subtitle stream to match the main video in size before overlaying
@example
 'scale2ref[b][a];[a][b]overlay'
@end example
@@ -9881,7 +9881,7 @@ stereo3d=sbsl:aybd
@end example

@item
-Convert input video from above below (left eye above, right eye below) to side by side crosseye.
+Convert input video from above bellow (left eye above, right eye below) to side by side crosseye.
@example
 stereo3d=abl:sbsr
@end example
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -205,10 +205,6 @@ For example to separate the fields with newlines and indention:
 ffprobe -dump_separator "
                          "  -i ~/videos/matrixbench_mpeg2.mpg
@end example
-
-@item max_streams @var{integer} (@emph{input})
-Specifies the maximum number of streams. This can be used to reject files that
-would require too many resources due to a large number of streams.
@end table

@c man end FORMAT OPTIONS
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -983,8 +983,8 @@ following image formats are supported:
@item Musepack SV8           @tab     @tab  X
@item Nellymoser Asao        @tab  X  @tab  X
@item On2 AVC (Audio for Video Codec) @tab     @tab  X
-@item Opus                   @tab  E  @tab  X
-    @tab encoding supported through external library libopus
+@item Opus                   @tab  E  @tab  E
+    @tab supported through external library libopus
@item PCM A-law              @tab  X  @tab  X
@item PCM mu-law             @tab  X  @tab  X
@item PCM signed 8-bit planar  @tab  X  @tab  X
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -53,7 +53,7 @@ Most distribution and operating system provide a package for it.
@section Cloning the source tree

@example
-git clone https://git.ffmpeg.org/ffmpeg.git <target>
+git clone git://source.ffmpeg.org/ffmpeg <target>
@end example

 This will put the FFmpeg sources into the directory @var{<target>}.
@@ -172,18 +172,11 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
@end example

-Make sure you have told Git your name, email address and GPG key
+Make sure you have told Git your name and email address

@example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
-git config --global user.signingkey ABCDEF0123245
-@end example
-
-Enable signing all commits or use -S
-
-@example
-git config --global commit.gpgsign true
@end example

 Use @option{--global} to set the global configuration for all your Git checkouts.
@@ -385,19 +378,6 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.


-@chapter gpg key generation
-
-If you have no gpg key yet, we recommend that you create a ed25519 based key as it
-is small, fast and secure. Especially it results in small signatures in git.
-
-@example
-gpg --default-new-key-algo "ed25519/cert,sign+cv25519/encr" --quick-generate-key "human@@server.com"
-@end example
-
-When generating a key, make sure the email specified matches the email used in git as some sites like
-github consider mismatches a reason to declare such commits unverified. After generating a key you
-can add it to the MAINTAINER file and upload it to a keyserver.
-
@chapter Pre-push checklist

 Once you have a set of commits that you feel are ready for pushing,
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -839,21 +839,6 @@ ffmpeg -i file.mpg -c copy \
     -y out.ts
@end example

-@section mxf, mxf_d10
-
-MXF muxer.
-
-@subsection Options
-
-The muxer options are:
-
-@table @option
-@item store_user_comments @var{bool}
-Set if user comments should be stored if available or never.
-IRT D-10 does not allow user comments. The default is thus to write them for
-mxf but not for mxf_d10
-@end table
-
@section null

 Null muxer.
--- a/doc/utils.texi
+++ b/doc/utils.texi
@@ -869,7 +869,7 @@ Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
 Return the maximum between @var{x} and @var{y}.

@item min(x, y)
-Return the minimum between @var{x} and @var{y}.
+Return the maximum between @var{x} and @var{y}.

@item mod(x, y)
 Compute the remainder of division of @var{x} by @var{y}.
--- a/doc/writing_filters.txt
+++ b/doc/writing_filters.txt
@@ -420,4 +420,4 @@ done:

 When all of this is done, you can submit your patch to the ffmpeg-devel
 mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.libera.chat.
+channel, #ffmpeg-devel on irc.freenode.net.
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -2396,12 +2396,8 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
        ist->dts = ist->next_dts;
        switch (ist->dec_ctx->codec_type) {
        case AVMEDIA_TYPE_AUDIO:
-            if (ist->dec_ctx->sample_rate) {
-                ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
-                                  ist->dec_ctx->sample_rate;
-            } else {
-                ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
-            }
+            ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
+                             ist->dec_ctx->sample_rate;
            break;
        case AVMEDIA_TYPE_VIDEO:
            if (ist->framerate.num) {
@@ -2451,9 +2447,6 @@ static void print_sdp(void)
        }
    }

-    if (!j)
-        goto fail;
-
    av_sdp_create(avc, j, sdp, sizeof(sdp));

    if (!sdp_filename) {
@@ -2469,7 +2462,6 @@ static void print_sdp(void)
        }
    }

-fail:
    av_freep(&avc);
 }

@@ -2886,8 +2878,7 @@ static int transcode_init(void)
             * overhead
             */
            if(!strcmp(oc->oformat->name, "avi")) {
-                if ( copy_tb<0 && ist->st->r_frame_rate.num
-                               && av_q2d(ist->st->r_frame_rate) >= av_q2d(ist->st->avg_frame_rate)
+                if ( copy_tb<0 && av_q2d(ist->st->r_frame_rate) >= av_q2d(ist->st->avg_frame_rate)
                               && 0.5/av_q2d(ist->st->r_frame_rate) > av_q2d(ist->st->time_base)
                               && 0.5/av_q2d(ist->st->r_frame_rate) > av_q2d(dec_ctx->time_base)
                               && av_q2d(ist->st->time_base) < 1.0/500 && av_q2d(dec_ctx->time_base) < 1.0/500
@@ -4122,8 +4113,6 @@ int main(int argc, char **argv)
    int ret;
    int64_t ti;

-    init_dynload();
-
    register_exit(ffmpeg_cleanup);

    setvbuf(stderr,NULL,_IONBF,0); /* win32 runtime needs this */
--- a/ffmpeg_opt.c
+++ b/ffmpeg_opt.c
@@ -2883,8 +2883,8 @@ enum OptGroup {
 };

 static const OptionGroupDef groups[] = {
-    [GROUP_OUTFILE] = { "output url",  NULL, OPT_OUTPUT },
-    [GROUP_INFILE]  = { "input url",   "i",  OPT_INPUT },
+    [GROUP_OUTFILE] = { "output file",  NULL, OPT_OUTPUT },
+    [GROUP_INFILE]  = { "input file",   "i",  OPT_INPUT },
 };

 static int open_files(OptionGroupList *l, const char *inout,
@@ -2903,7 +2903,6 @@ static int open_files(OptionGroupList *l, const char *inout,
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error parsing options for %s file "
                   "%s.\n", inout, g->arg);
-            uninit_options(&o);
            return ret;
        }

--- a/ffplay.c
+++ b/ffplay.c
@@ -2869,7 +2869,7 @@ static int read_thread(void *arg)
        AVStream *st = ic->streams[i];
        enum AVMediaType type = st->codec->codec_type;
        st->discard = AVDISCARD_ALL;
-        if (type >= 0 && wanted_stream_spec[type] && st_index[type] == -1)
+        if (wanted_stream_spec[type] && st_index[type] == -1)
            if (avformat_match_stream_specifier(ic, st, wanted_stream_spec[type]) > 0)
                st_index[type] = i;
    }
@@ -3684,8 +3684,6 @@ int main(int argc, char **argv)
    VideoState *is;
    char dummy_videodriver[] = "SDL_VIDEODRIVER=dummy";

-    init_dynload();
-
    av_log_set_flags(AV_LOG_SKIP_REPEATED);
    parse_loglevel(argc, argv, options);

--- a/ffprobe.c
+++ b/ffprobe.c
@@ -1732,57 +1732,6 @@ static inline int show_tags(WriterContext *w, AVDictionary *tags, int section_id
    return ret;
 }

-
-static void print_color_range(WriterContext *w, enum AVColorRange color_range, const char *fallback)
-{
-    const char *val = av_color_range_name(color_range);
-    if (!val || color_range == AVCOL_RANGE_UNSPECIFIED) {
-        print_str_opt("color_range", fallback);
-    } else {
-        print_str("color_range", val);
-    }
-}
-
-static void print_color_space(WriterContext *w, enum AVColorSpace color_space)
-{
-    const char *val = av_color_space_name(color_space);
-    if (!val || color_space == AVCOL_SPC_UNSPECIFIED) {
-        print_str_opt("color_space", "unknown");
-    } else {
-        print_str("color_space", val);
-    }
-}
-
-static void print_primaries(WriterContext *w, enum AVColorPrimaries color_primaries)
-{
-    const char *val = av_color_primaries_name(color_primaries);
-    if (!val || color_primaries == AVCOL_PRI_UNSPECIFIED) {
-        print_str_opt("color_primaries", "unknown");
-    } else {
-        print_str("color_primaries", val);
-    }
-}
-
-static void print_color_trc(WriterContext *w, enum AVColorTransferCharacteristic color_trc)
-{
-    const char *val = av_color_transfer_name(color_trc);
-    if (!val || color_trc == AVCOL_TRC_UNSPECIFIED) {
-        print_str_opt("color_transfer", "unknown");
-    } else {
-        print_str("color_transfer", val);
-    }
-}
-
-static void print_chroma_location(WriterContext *w, enum AVChromaLocation chroma_location)
-{
-    const char *val = av_chroma_location_name(chroma_location);
-    if (!val || chroma_location == AVCHROMA_LOC_UNSPECIFIED) {
-        print_str_opt("chroma_location", "unspecified");
-    } else {
-        print_str("chroma_location", val);
-    }
-}
-
 static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pkt, int packet_idx)
 {
    char val_str[128];
@@ -2225,12 +2174,28 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
            if (s) print_str    ("pix_fmt", s);
            else   print_str_opt("pix_fmt", "unknown");
            print_int("level",   dec_ctx->level);
+            if (dec_ctx->color_range != AVCOL_RANGE_UNSPECIFIED)
+                print_str    ("color_range", av_color_range_name(dec_ctx->color_range));
+            else
+                print_str_opt("color_range", "N/A");
+            s = av_get_colorspace_name(dec_ctx->colorspace);
+            if (s) print_str    ("color_space", s);
+            else   print_str_opt("color_space", "unknown");

-            print_color_range(w, dec_ctx->color_range, "N/A");
-            print_color_space(w, dec_ctx->colorspace);
-            print_color_trc(w, dec_ctx->color_trc);
-            print_primaries(w, dec_ctx->color_primaries);
-            print_chroma_location(w, dec_ctx->chroma_sample_location);
+            if (dec_ctx->color_trc != AVCOL_TRC_UNSPECIFIED)
+                print_str("color_transfer", av_color_transfer_name(dec_ctx->color_trc));
+            else
+                print_str_opt("color_transfer", av_color_transfer_name(dec_ctx->color_trc));
+
+            if (dec_ctx->color_primaries != AVCOL_PRI_UNSPECIFIED)
+                print_str("color_primaries", av_color_primaries_name(dec_ctx->color_primaries));
+            else
+                print_str_opt("color_primaries", av_color_primaries_name(dec_ctx->color_primaries));
+
+            if (dec_ctx->chroma_sample_location != AVCHROMA_LOC_UNSPECIFIED)
+                print_str("chroma_location", av_chroma_location_name(dec_ctx->chroma_sample_location));
+            else
+                print_str_opt("chroma_location", av_chroma_location_name(dec_ctx->chroma_sample_location));

            if (dec_ctx->timecode_frame_start >= 0) {
                char tcbuf[AV_TIMECODE_STR_SIZE];
@@ -3177,8 +3142,6 @@ int main(int argc, char **argv)
    char *w_name = NULL, *w_args = NULL;
    int ret, i;

-    init_dynload();
-
    av_log_set_flags(AV_LOG_SKIP_REPEATED);
    register_exit(ffprobe_cleanup);

--- a/ffserver.c
+++ b/ffserver.c
@@ -2580,10 +2580,8 @@ static int http_receive_data(HTTPContext *c)
        } else if (c->buffer_ptr - c->buffer >= 2 &&
                   !memcmp(c->buffer_ptr - 1, "\r\n", 2)) {
            c->chunk_size = strtol(c->buffer, 0, 16);
-            if (c->chunk_size <= 0) { // end of stream or invalid chunk size
-                c->chunk_size = 0;
+            if (c->chunk_size == 0) // end of stream
                goto fail;
-            }
            c->buffer_ptr = c->buffer;
            break;
        } else if (++loop_run > 10)
@@ -2605,7 +2603,6 @@ static int http_receive_data(HTTPContext *c)
            /* end of connection : close it */
            goto fail;
        else {
-            av_assert0(len <= c->chunk_size);
            c->chunk_size -= len;
            c->buffer_ptr += len;
            c->data_count += len;
@@ -3820,8 +3817,6 @@ int main(int argc, char **argv)
    struct sigaction sigact = { { 0 } };
    int ret = 0;

-    init_dynload();
-
    config.filename = av_strdup("/etc/ffserver.conf");

    parse_loglevel(argc, argv, options);
--- a/libavcodec/012v.c
+++ b/libavcodec/012v.c
@@ -131,8 +131,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
            u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
            v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
            memcpy(y, y_temp, sizeof(*y) * (width - x));
-            memcpy(u, u_temp, sizeof(*u) * ((width - x + 1) / 2));
-            memcpy(v, v_temp, sizeof(*v) * ((width - x + 1) / 2));
+            memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
+            memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
        }

        line_end += stride;
--- a/libavcodec/4xm.c
+++ b/libavcodec/4xm.c
@@ -158,7 +158,7 @@ typedef struct FourXContext {
 #define FIX_1_847759065 121095
 #define FIX_2_613125930 171254

-#define MULTIPLY(var, const) ((int)((var) * (unsigned)(const)) >> 16)
+#define MULTIPLY(var, const) (((var) * (const)) >> 16)

 static void idct(int16_t block[64])
 {
@@ -351,8 +351,6 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, const uint16_t *src,
    index = size2index[log2h][log2w];
    av_assert0(index >= 0);

-    if (get_bits_left(&f->gb) < 1)
-        return AVERROR_INVALIDDATA;
    h     = 1 << log2h;
    code  = get_vlc2(&f->gb, block_type_vlc[1 - (f->version > 1)][index].table,
                     BLOCK_TYPE_VLC_BITS, 1);
@@ -498,9 +496,9 @@ static int decode_i_block(FourXContext *f, int16_t *block)
 {
    int code, i, j, level, val;

-    if (get_bits_left(&f->pre_gb) < 2) {
-        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->pre_gb));
-        return AVERROR_INVALIDDATA;
+    if (get_bits_left(&f->gb) < 2){
+        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->gb));
+        return -1;
    }

    /* DC coef */
@@ -525,10 +523,6 @@ static int decode_i_block(FourXContext *f, int16_t *block)
            break;
        if (code == 0xf0) {
            i += 16;
-            if (i >= 64) {
-                av_log(f->avctx, AV_LOG_ERROR, "run %d overflow\n", i);
-                return 0;
-            }
        } else {
            if (code & 0xf) {
                level = get_xbits(&f->gb, code & 0xf);
@@ -738,7 +732,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
        for (x = 0; x < width; x += 16) {
            unsigned int color[4] = { 0 }, bits;
            if (buf_end - buf < 8)
-                return AVERROR_INVALIDDATA;
+                return -1;
            // warning following is purely guessed ...
            color[0] = bytestream2_get_le16u(&g3);
            color[1] = bytestream2_get_le16u(&g3);
--- a/libavcodec/8bps.c
+++ b/libavcodec/8bps.c
@@ -70,9 +70,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    unsigned char *planemap = c->planemap;
    int ret;

-    if (buf_size < planes * height *2)
-        return AVERROR_INVALIDDATA;
-
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;

@@ -123,15 +120,12 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    }

    if (avctx->bits_per_coded_sample <= 8) {
-        int size;
        const uint8_t *pal = av_packet_get_side_data(avpkt,
                                                     AV_PKT_DATA_PALETTE,
-                                                     &size);
-        if (pal && size == AVPALETTE_SIZE) {
+                                                     NULL);
+        if (pal) {
            frame->palette_has_changed = 1;
            memcpy(c->pal, pal, AVPALETTE_SIZE);
-        } else if (pal) {
-            av_log(avctx, AV_LOG_ERROR, "Palette size %d is wrong\n", size);
        }

        memcpy (frame->data[1], c->pal, AVPALETTE_SIZE);
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -861,7 +861,7 @@ OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
                                          mpeg4videodec.o mpeg4video.o \
-                                          ituh263dec.o h263dec.o h263data.o
+                                          ituh263dec.o h263dec.o
 OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                          mpegaudiodecheader.o mpegaudiodata.o
--- a/libavcodec/aac_defines.h
+++ b/libavcodec/aac_defines.h
@@ -45,7 +45,7 @@
 #define Q30(x)              (int)((x)*1073741824.0 + 0.5)
 #define Q31(x)              (int)((x)*2147483648.0 + 0.5)
 #define RANGE15(x)          x
-#define GET_GAIN(x, y)      (-(y) * (1 << (x))) + 1024
+#define GET_GAIN(x, y)      (-(y) << (x)) + 1024
 #define AAC_MUL16(x, y)     (int)(((int64_t)(x) * (y) + 0x8000) >> 16)
 #define AAC_MUL26(x, y)     (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
 #define AAC_MUL30(x, y)     (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
@@ -72,7 +72,7 @@
 #define AAC_MSUB31_V3(x, y, z)    (int)((((int64_t)(x) * (z)) - \
                                      ((int64_t)(y) * (z)) + \
                                        0x40000000) >> 31)
-#define AAC_HALF_SUM(x, y)  (((x) >> 1) + ((y) >> 1))
+#define AAC_HALF_SUM(x, y)  (x) >> 1 + (y) >> 1
 #define AAC_SRA_R(x, y)     (int)(((x) + (1 << ((y) - 1))) >> (y))

 #else
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -499,7 +499,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
    }
    while (idx) {
        sce->sf_idx[bandaddr[idx]] = minq + q0;
-        minq = FFMAX(paths[idx][minq].prev, 0);
+        minq = paths[idx][minq].prev;
        idx--;
    }
    //set the same quantizers inside window groups
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -403,8 +403,6 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
            } else {
                int esc;
                do {
-                    if (get_bits_left(gb) < 9)
-                        return AVERROR_INVALIDDATA;
                    esc = get_bits(gb, 1);
                    skip_bits(gb, 8);
                } while (esc);
@@ -425,8 +423,6 @@ static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
    if (ctx->frame_length_type == 0) {
        int mux_slot_length = 0;
        do {
-            if (get_bits_left(gb) < 8)
-                return AVERROR_INVALIDDATA;
            tmp = get_bits(gb, 8);
            mux_slot_length += tmp;
        } while (tmp == 255);
@@ -456,7 +452,7 @@ static int read_audio_mux_element(struct LATMContext *latmctx,
    }
    if (latmctx->audio_mux_version_A == 0) {
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
-        if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
+        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
            return AVERROR_INVALIDDATA;
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
--- a/libavcodec/aacdec_fixed.c
+++ b/libavcodec/aacdec_fixed.c
@@ -124,7 +124,7 @@ static inline int *DEC_SQUAD(int *dst, unsigned idx)
 static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
 {
    dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
-    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
+    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) << 1));

    return dst + 2;
 }
@@ -133,16 +133,16 @@ static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
 {
    unsigned nz = idx >> 12;

-    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
+    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) << 1));
    sign <<= nz & 1;
    nz >>= 1;
-    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
+    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
    sign <<= nz & 1;
    nz >>= 1;
-    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
+    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
    sign <<= nz & 1;
    nz >>= 1;
-    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
+    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));

    return dst + 4;
 }
@@ -154,9 +154,9 @@ static void vector_pow43(int *coefs, int len)
    for (i=0; i<len; i++) {
        coef = coefs[i];
        if (coef < 0)
-            coef = -(int)cbrt_tab[(-coef) & 8191];
+            coef = -(int)cbrt_tab[-coef];
        else
-            coef =  (int)cbrt_tab[  coef  & 8191];
+            coef = (int)cbrt_tab[coef];
        coefs[i] = coef;
    }
 }
@@ -170,36 +170,31 @@ static void subband_scale(int *dst, int *src, int scale, int offset, int len)

    s = offset - (s >> 2);

-    if (s > 31) {
-        for (i=0; i<len; i++) {
-            dst[i] = 0;
-        }
-    } else if (s > 0) {
+    if (s > 0) {
        round = 1 << (s-1);
        for (i=0; i<len; i++) {
            out = (int)(((int64_t)src[i] * c) >> 32);
            dst[i] = ((int)(out+round) >> s) * ssign;
        }
-    } else if (s > -32) {
+    }
+    else {
        s = s + 32;
        round = 1 << (s-1);
        for (i=0; i<len; i++) {
            out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
-            dst[i] = out * (unsigned)ssign;
+            dst[i] = out * ssign;
        }
-    } else {
-        av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n");
    }
 }

 static void noise_scale(int *coefs, int scale, int band_energy, int len)
 {
-    int s = -scale;
+    int ssign = scale < 0 ? -1 : 1;
+    int s = FFABS(scale);
    unsigned int round;
    int i, out, c = exp2tab[s & 3];
    int nlz = 0;

-    av_assert0(s >= 0);
    while (band_energy > 0x7fff) {
        band_energy >>= 1;
        nlz++;
@@ -207,28 +202,19 @@ static void noise_scale(int *coefs, int scale, int band_energy, int len)
    c /= band_energy;
    s = 21 + nlz - (s >> 2);

-    if (s > 31) {
-        for (i=0; i<len; i++) {
-            coefs[i] = 0;
-        }
-    } else if (s >= 0) {
-        round = s ? 1 << (s-1) : 0;
+    if (s > 0) {
+        round = 1 << (s-1);
        for (i=0; i<len; i++) {
            out = (int)(((int64_t)coefs[i] * c) >> 32);
-            coefs[i] = -((int)(out+round) >> s);
+            coefs[i] = ((int)(out+round) >> s) * ssign;
        }
    }
    else {
        s = s + 32;
-        if (s > 0) {
-            round = 1 << (s-1);
-            for (i=0; i<len; i++) {
-                out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
-                coefs[i] = -out;
-            }
-        } else {
-            for (i=0; i<len; i++)
-                coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
+        round = 1 << (s-1);
+        for (i=0; i<len; i++) {
+            out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
+            coefs[i] = out * ssign;
        }
    }
 }
@@ -309,12 +295,8 @@ static av_always_inline void predict(PredictorState *ps, int *coef,
    if (output_enable) {
        int shift = 28 - pv.exp;

-        if (shift < 31) {
-            if (shift > 0) {
-                *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
-            } else
-                *coef += (unsigned)pv.mant << -shift;
-        }
+        if (shift < 31)
+            *coef += (pv.mant + (1 << (shift - 1))) >> shift;
    }

    e0 = av_int2sf(*coef, 2);
@@ -379,9 +361,7 @@ static void apply_dependent_coupling_fixed(AACContext *ac,
                    shift = (gain-1024) >> 3;
                }

-                if (shift < -31) {
-                    // Nothing to do
-                } else if (shift < 0) {
+                if (shift < 0) {
                    shift = -shift;
                    round = 1 << (shift - 1);

@@ -389,7 +369,7 @@ static void apply_dependent_coupling_fixed(AACContext *ac,
                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
                                       (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
+                            dest[group * 128 + k] += (tmp + round) >> shift;
                        }
                    }
                }
@@ -398,7 +378,7 @@ static void apply_dependent_coupling_fixed(AACContext *ac,
                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
                                        (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += tmp * (1U << shift);
+                            dest[group * 128 + k] += tmp << shift;
                        }
                    }
                }
@@ -421,7 +401,7 @@ static void apply_independent_coupling_fixed(AACContext *ac,
    int i, c, shift, round, tmp;
    const int gain = cce->coup.gain[index][0];
    const int *src = cce->ch[0].ret;
-    unsigned int *dest = target->ret;
+    int *dest = target->ret;
    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);

    c = cce_scale_fixed[gain & 7];
@@ -438,7 +418,7 @@ static void apply_independent_coupling_fixed(AACContext *ac,
    else {
      for (i = 0; i < len; i++) {
          tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
-          dest[i] += tmp * (1U << shift);
+          dest[i] += tmp << shift;
      }
    }
 }
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
@@ -449,10 +449,6 @@ static int output_configure(AACContext *ac,
        int type =         layout_map[i][0];
        int id =           layout_map[i][1];
        id_map[type][id] = type_counts[type]++;
-        if (id_map[type][id] >= MAX_ELEM_ID) {
-            avpriv_request_sample(ac->avctx, "Remapped id too large\n");
-            return AVERROR_PATCHWELCOME;
-        }
    }
    // Try to sniff a reasonable channel order, otherwise output the
    // channels in the order the PCE declared them.
@@ -1243,8 +1239,6 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
    const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
    const int aot = m4ac->object_type;
    const int sampling_index = m4ac->sampling_index;
-    int ret_fail = AVERROR_INVALIDDATA;
-
    if (aot != AOT_ER_AAC_ELD) {
        if (get_bits1(gb)) {
            av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
@@ -1295,10 +1289,8 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                ics->num_swb       =    ff_aac_num_swb_512[sampling_index];
                ics->tns_max_bands =  ff_tns_max_bands_512[sampling_index];
            }
-            if (!ics->num_swb || !ics->swb_offset) {
-                ret_fail = AVERROR_BUG;
-                goto fail;
-            }
+            if (!ics->num_swb || !ics->swb_offset)
+                return AVERROR_BUG;
        } else {
            ics->swb_offset    =    ff_swb_offset_1024[sampling_index];
            ics->num_swb       =   ff_aac_num_swb_1024[sampling_index];
@@ -1322,8 +1314,7 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                if (aot == AOT_ER_AAC_LD) {
                    av_log(ac->avctx, AV_LOG_ERROR,
                           "LTP in ER AAC LD not yet implemented.\n");
-                    ret_fail = AVERROR_PATCHWELCOME;
-                    goto fail;
+                    return AVERROR_PATCHWELCOME;
                }
                if ((ics->ltp.present = get_bits(gb, 1)))
                    decode_ltp(&ics->ltp, gb, ics->max_sfb);
@@ -1342,7 +1333,7 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
    return 0;
 fail:
    ics->max_sfb = 0;
-    return ret_fail;
+    return AVERROR_INVALIDDATA;
 }

 /**
@@ -1602,24 +1593,25 @@ static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
                }
            } else if (cbt_m1 == NOISE_BT - 1) {
                for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+#if !USE_FIXED
+                    float scale;
+#endif /* !USE_FIXED */
                    INTFLOAT band_energy;
-#if USE_FIXED
+
                    for (k = 0; k < off_len; k++) {
                        ac->random_state  = lcg_random(ac->random_state);
+#if USE_FIXED
                        cfo[k] = ac->random_state >> 3;
+#else
+                        cfo[k] = ac->random_state;
+#endif /* USE_FIXED */
                    }

+#if USE_FIXED
                    band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
                    band_energy = fixed_sqrt(band_energy, 31);
                    noise_scale(cfo, sf[idx], band_energy, off_len);
 #else
-                    float scale;
-
-                    for (k = 0; k < off_len; k++) {
-                        ac->random_state  = lcg_random(ac->random_state);
-                        cfo[k] = ac->random_state;
-                    }
-
                    band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
                    scale = sf[idx] / sqrtf(band_energy);
                    ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
@@ -1928,17 +1920,16 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
    global_gain = get_bits(gb, 8);

    if (!common_window && !scale_flag) {
-        ret = decode_ics_info(ac, ics, gb);
-        if (ret < 0)
-            goto fail;
+        if (decode_ics_info(ac, ics, gb) < 0)
+            return AVERROR_INVALIDDATA;
    }

    if ((ret = decode_band_types(ac, sce->band_type,
                                 sce->band_type_run_end, gb, ics)) < 0)
-        goto fail;
+        return ret;
    if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
                                  sce->band_type, sce->band_type_run_end)) < 0)
-        goto fail;
+        return ret;

    pulse_present = 0;
    if (!scale_flag) {
@@ -1946,48 +1937,37 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
                av_log(ac->avctx, AV_LOG_ERROR,
                       "Pulse tool not allowed in eight short sequence.\n");
-                ret = AVERROR_INVALIDDATA;
-                goto fail;
+                return AVERROR_INVALIDDATA;
            }
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
                av_log(ac->avctx, AV_LOG_ERROR,
                       "Pulse data corrupt or invalid.\n");
-                ret = AVERROR_INVALIDDATA;
-                goto fail;
+                return AVERROR_INVALIDDATA;
            }
        }
        tns->present = get_bits1(gb);
-        if (tns->present && !er_syntax) {
-            ret = decode_tns(ac, tns, gb, ics);
-            if (ret < 0)
-                goto fail;
-        }
+        if (tns->present && !er_syntax)
+            if (decode_tns(ac, tns, gb, ics) < 0)
+                return AVERROR_INVALIDDATA;
        if (!eld_syntax && get_bits1(gb)) {
            avpriv_request_sample(ac->avctx, "SSR");
-            ret = AVERROR_PATCHWELCOME;
-            goto fail;
+            return AVERROR_PATCHWELCOME;
        }
        // I see no textual basis in the spec for this occurring after SSR gain
        // control, but this is what both reference and real implmentations do
-        if (tns->present && er_syntax) {
-            ret = decode_tns(ac, tns, gb, ics);
-            if (ret < 0)
-                goto fail;
-        }
+        if (tns->present && er_syntax)
+            if (decode_tns(ac, tns, gb, ics) < 0)
+                return AVERROR_INVALIDDATA;
    }

-    ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
-                                    &pulse, ics, sce->band_type);
-    if (ret < 0)
-        goto fail;
+    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
+                                    &pulse, ics, sce->band_type) < 0)
+        return AVERROR_INVALIDDATA;

    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
        apply_prediction(ac, sce);

    return 0;
-fail:
-    tns->present = 0;
-    return ret;
 }

 /**
@@ -2159,11 +2139,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);

    sign  = get_bits(gb, 1);
-#if USE_FIXED
-    scale = get_bits(gb, 2);
-#else
-    scale = cce_scale[get_bits(gb, 2)];
-#endif
+    scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];

    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
        return ret;
@@ -2177,10 +2153,6 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
            gain_cache = GET_GAIN(scale, gain);
-#if USE_FIXED
-            if ((abs(gain_cache)-1024) >> 3 > 30)
-                return AVERROR(ERANGE);
-#endif
        }
        if (coup->coupling_point == AFTER_IMDCT) {
            coup->gain[c][0] = gain_cache;
@@ -2198,10 +2170,6 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
                                    t >>= 1;
                                }
                                gain_cache = GET_GAIN(scale, t) * s;
-#if USE_FIXED
-                                if ((abs(gain_cache)-1024) >> 3 > 30)
-                                    return AVERROR(ERANGE);
-#endif
                            }
                        }
                        coup->gain[c][idx] = gain_cache;
@@ -2384,9 +2352,6 @@ static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
    INTFLOAT lpc[TNS_MAX_ORDER];
    INTFLOAT tmp[TNS_MAX_ORDER+1];

-    if(!mmm)
-        return;
-
    for (w = 0; w < ics->num_windows; w++) {
        bottom = ics->num_swb;
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
@@ -2551,7 +2516,7 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
        ac->mdct.imdct_half(&ac->mdct, buf, in);
 #if USE_FIXED
        for (i=0; i<1024; i++)
-          buf[i] = (buf[i] + 4LL) >> 3;
+          buf[i] = (buf[i] + 4) >> 3;
 #endif /* USE_FIXED */
    }

@@ -2924,11 +2889,6 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,

    spectral_to_sample(ac, samples);

-    if (!ac->frame->data[0] && samples) {
-        av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
-        return AVERROR_INVALIDDATA;
-    }
-
    ac->frame->nb_samples = samples;
    ac->frame->sample_rate = avctx->sample_rate;
    *got_frame_ptr = 1;
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -493,7 +493,6 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
    int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
    int chan_el_counter[4];
    FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
-    int k;

    if (s->last_frame == 2)
        return 0;
@@ -548,7 +547,6 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            ics->num_windows        = wi[ch].num_windows;
            ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
            ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
-            ics->max_sfb            = FFMIN(ics->max_sfb, ics->num_swb);
            ics->swb_offset         = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
                                        ff_swb_offset_128 [s->samplerate_index]:
                                        ff_swb_offset_1024[s->samplerate_index];
@@ -573,12 +571,9 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            }

            apply_window_and_mdct(s, &cpe->ch[ch], overlap);
-
-            for (k = 0; k < 1024; k++) {
-                if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
-                    av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
-                    return AVERROR(EINVAL);
-                }
+            if (isnan(cpe->ch->coeffs[0])) {
+                av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
+                return AVERROR(EINVAL);
            }
            avoid_clipping(s, &cpe->ch[ch]);
        }
--- a/libavcodec/aacps.c
+++ b/libavcodec/aacps.c
@@ -499,13 +499,13 @@ static void map_idx_34_to_20(int8_t *par_mapped, const int8_t *par, int full)
 static void map_val_34_to_20(INTFLOAT par[PS_MAX_NR_IIDICC])
 {
 #if USE_FIXED
-    par[ 0] = (int)(((int64_t)(par[ 0] + (unsigned)(par[ 1]>>1)) * 1431655765 + \
+    par[ 0] = (int)(((int64_t)(par[ 0] + (par[ 1]>>1)) * 1431655765 + \
                      0x40000000) >> 31);
-    par[ 1] = (int)(((int64_t)((par[ 1]>>1) + (unsigned)par[ 2]) * 1431655765 + \
+    par[ 1] = (int)(((int64_t)((par[ 1]>>1) + par[ 2]) * 1431655765 + \
                      0x40000000) >> 31);
-    par[ 2] = (int)(((int64_t)(par[ 3] + (unsigned)(par[ 4]>>1)) * 1431655765 + \
+    par[ 2] = (int)(((int64_t)(par[ 3] + (par[ 4]>>1)) * 1431655765 + \
                      0x40000000) >> 31);
-    par[ 3] = (int)(((int64_t)((par[ 4]>>1) + (unsigned)par[ 5]) * 1431655765 + \
+    par[ 3] = (int)(((int64_t)((par[ 4]>>1) + par[ 5]) * 1431655765 + \
                      0x40000000) >> 31);
 #else
    par[ 0] = (2*par[ 0] +   par[ 1]) * 0.33333333f;
@@ -692,17 +692,26 @@ static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT
    for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
        for (n = n0; n < nL; n++) {
            int decayed_peak;
+            int denom;
+
            decayed_peak = (int)(((int64_t)peak_decay_factor * \
                                           peak_decay_nrg[i] + 0x40000000) >> 31);
            peak_decay_nrg[i] = FFMAX(decayed_peak, power[i][n]);
-            power_smooth[i] += (power[i][n] + 2LL - power_smooth[i]) >> 2;
-            peak_decay_diff_smooth[i] += (peak_decay_nrg[i] + 2LL - power[i][n] - \
-                                          peak_decay_diff_smooth[i]) >> 2;
-
-            if (peak_decay_diff_smooth[i]) {
-                transient_gain[i][n] = FFMIN(power_smooth[i]*43691LL / peak_decay_diff_smooth[i], 1<<16);
-            } else
-                transient_gain[i][n] = 1 << 16;
+            power_smooth[i] += (power[i][n] - power_smooth[i] + 2) >> 2;
+            peak_decay_diff_smooth[i] += (peak_decay_nrg[i] - power[i][n] - \
+                                          peak_decay_diff_smooth[i] + 2) >> 2;
+            denom = peak_decay_diff_smooth[i] + (peak_decay_diff_smooth[i] >> 1);
+            if (denom > power_smooth[i]) {
+              int p = power_smooth[i];
+              while (denom < 0x40000000) {
+                denom <<= 1;
+                p <<= 1;
+              }
+              transient_gain[i][n] = p / (denom >> 16);
+            }
+            else {
+              transient_gain[i][n] = 1 << 16;
+            }
        }
    }
 #else
@@ -931,9 +940,9 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
            LOCAL_ALIGNED_16(INTFLOAT, h_step, [2], [4]);
            int start = ps->border_position[e];
            int stop  = ps->border_position[e+1];
-            INTFLOAT width = Q30(1.f) / ((stop - start) ? (stop - start) : 1);
+            INTFLOAT width = Q30(1.f) / (stop - start);
 #if USE_FIXED
-            width = FFMIN(2U*width, INT_MAX);
+            width <<= 1;
 #endif
            b = k_to_i[k];
            h[0][0] = H11[0][e][b];
@@ -966,7 +975,7 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
                h_step[1][3] = AAC_MSUB31_V3(H22[1][e+1][b], h[1][3], width);
            }
            ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
-                l[k] + 1 + start, r[k] + 1 + start,
+                l[k] + start + 1, r[k] + start + 1,
                h, h_step, stop - start);
        }
    }
--- a/libavcodec/aacpsdsp_template.c
+++ b/libavcodec/aacpsdsp_template.c
@@ -54,10 +54,10 @@ static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
        INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];

        for (j = 0; j < 6; j++) {
-            INT64FLOAT in0_re = in[j][0];
-            INT64FLOAT in0_im = in[j][1];
-            INT64FLOAT in1_re = in[12-j][0];
-            INT64FLOAT in1_im = in[12-j][1];
+            INTFLOAT in0_re = in[j][0];
+            INTFLOAT in0_im = in[j][1];
+            INTFLOAT in1_re = in[12-j][0];
+            INTFLOAT in1_im = in[12-j][1];
            sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
                      (INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
            sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -303,9 +303,6 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
    const int bandwidth    = ctx->avctx->cutoff ? ctx->avctx->cutoff : AAC_CUTOFF(ctx->avctx);
    const float num_bark   = calc_bark((float)bandwidth);

-    if (bandwidth <= 0)
-        return AVERROR(EINVAL);
-
    ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
    if (!ctx->model_priv_data)
        return AVERROR(ENOMEM);
@@ -750,7 +747,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,

        if (pe < 1.15f * desired_pe) {
            /* 6.6.1.3.6 "Final threshold modification by linearization" */
-            norm_fac = norm_fac ? 1.0f / norm_fac : 0;
+            norm_fac = 1.0f / norm_fac;
            for (w = 0; w < wi->num_windows*16; w += 16) {
                for (g = 0; g < num_bands; g++) {
                    AacPsyBand *band = &pch->band[w+g];
--- a/libavcodec/aacsbr_fixed.c
+++ b/libavcodec/aacsbr_fixed.c
@@ -169,10 +169,6 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
                else
                  temp1.mant = 0x20000000;
                temp1.exp = (temp1.exp >> 1) + 1;
-                if (temp1.exp > 66) { // temp1 > 1E20
-                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
-                    temp1 = FLOAT_1;
-                }

                temp2.exp = (pan_offset - sbr->data[1].env_facs[e][k].mant) * alpha;
                if (temp2.exp & 1)
@@ -192,10 +188,6 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
                temp1.exp = NOISE_FLOOR_OFFSET - \
                    sbr->data[0].noise_facs[e][k].mant + 2;
                temp1.mant = 0x20000000;
-                if (temp1.exp > 66) { // temp1 > 1E20
-                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
-                    temp1 = FLOAT_1;
-                }
                temp2.exp = 12 - sbr->data[1].noise_facs[e][k].mant + 1;
                temp2.mant = 0x20000000;
                fac   = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
@@ -216,10 +208,7 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
                    else
                        temp1.mant = 0x20000000;
                    temp1.exp = (temp1.exp >> 1) + 1;
-                    if (temp1.exp > 66) { // temp1 > 1E20
-                        av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
-                        temp1 = FLOAT_1;
-                    }
+
                    sbr->data[ch].env_facs[e][k] = temp1;
                }
            for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
@@ -291,8 +280,6 @@ static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
        shift = a00.exp;
        if (shift >= 3)
            alpha0[k][0] = 0x7fffffff;
-        else if (shift <= -30)
-            alpha0[k][0] = 0;
        else {
            a00.mant <<= 1;
            shift = 2-shift;
@@ -307,8 +294,6 @@ static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
        shift = a01.exp;
        if (shift >= 3)
            alpha0[k][1] = 0x7fffffff;
-        else if (shift <= -30)
-            alpha0[k][1] = 0;
        else {
            a01.mant <<= 1;
            shift = 2-shift;
@@ -322,8 +307,6 @@ static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
        shift = a10.exp;
        if (shift >= 3)
            alpha1[k][0] = 0x7fffffff;
-        else if (shift <= -30)
-            alpha1[k][0] = 0;
        else {
            a10.mant <<= 1;
            shift = 2-shift;
@@ -338,8 +321,6 @@ static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
        shift = a11.exp;
        if (shift >= 3)
            alpha1[k][1] = 0x7fffffff;
-        else if (shift <= -30)
-            alpha1[k][1] = 0;
        else {
            a11.mant <<= 1;
            shift = 2-shift;
@@ -416,8 +397,7 @@ static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
        int delta = !((e == e_a[1]) || (e == e_a[0]));
        for (k = 0; k < sbr->n_lim; k++) {
            SoftFloat gain_boost, gain_max;
-            SoftFloat sum[2];
-            sum[0] = sum[1] = FLOAT_0;
+            SoftFloat sum[2] = { { 0, 0}, { 0, 0 } };
            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
                const SoftFloat temp = av_div_sf(sbr->e_origmapped[e][m],
                                            av_add_sf(FLOAT_1, sbr->q_mapped[e][m]));
@@ -440,7 +420,6 @@ static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
                                                av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
                                                av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
                }
-                sbr->gain[e][m] = av_add_sf(sbr->gain[e][m], FLOAT_MIN);
            }
            for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
                sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
@@ -574,39 +553,25 @@ static void sbr_hf_assemble(int Y1[38][64][2],
                int idx = indexsine&1;
                int A = (1-((indexsine+(kx & 1))&2));
                int B = (A^(-idx)) + idx;
-                unsigned *out = &Y1[i][kx][idx];
-                int shift;
-                unsigned round;
+                int *out = &Y1[i][kx][idx];
+                int shift, round;

                SoftFloat *in  = sbr->s_m[e];
                for (m = 0; m+1 < m_max; m+=2) {
-                    int shift2;
-                    shift = 22 - in[m  ].exp;
-                    shift2= 22 - in[m+1].exp;
-                    if (shift < 1 || shift2 < 1) {
-                        av_log(NULL, AV_LOG_ERROR, "Overflow in sbr_hf_assemble, shift=%d,%d\n", shift, shift2);
-                        return;
-                    }
-                    if (shift < 32) {
-                        round = 1 << (shift-1);
-                        out[2*m  ] += (int)(in[m  ].mant * A + round) >> shift;
-                    }
+                  shift = 22 - in[m  ].exp;
+                  round = 1 << (shift-1);
+                  out[2*m  ] += (in[m  ].mant * A + round) >> shift;

-                    if (shift2 < 32) {
-                        round = 1 << (shift2-1);
-                        out[2*m+2] += (int)(in[m+1].mant * B + round) >> shift2;
-                    }
+                  shift = 22 - in[m+1].exp;
+                  round = 1 << (shift-1);
+                  out[2*m+2] += (in[m+1].mant * B + round) >> shift;
                }
                if(m_max&1)
                {
-                    shift = 22 - in[m  ].exp;
-                    if (shift < 1) {
-                        av_log(NULL, AV_LOG_ERROR, "Overflow in sbr_hf_assemble, shift=%d\n", shift);
-                        return;
-                    } else if (shift < 32) {
-                        round = 1 << (shift-1);
-                        out[2*m  ] += (int)(in[m  ].mant * A + round) >> shift;
-                    }
+                  shift = 22 - in[m  ].exp;
+                  round = 1 << (shift-1);
+
+                  out[2*m  ] += (in[m  ].mant * A + round) >> shift;
                }
            }
            indexnoise = (indexnoise + m_max) & 0x1ff;
--- a/libavcodec/aacsbr_template.c
+++ b/libavcodec/aacsbr_template.c
@@ -70,7 +70,6 @@ av_cold void AAC_RENAME(ff_aac_sbr_init)(void)
 /** Places SBR in pure upsampling mode. */
 static void sbr_turnoff(SpectralBandReplication *sbr) {
    sbr->start = 0;
-    sbr->ready_for_dequant = 0;
    // Init defults used in pure upsampling mode
    sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
    sbr->m[1] = 0;
@@ -178,7 +177,6 @@ static unsigned int read_sbr_header(SpectralBandReplication *sbr, GetBitContext
    SpectrumParameters old_spectrum_params;

    sbr->start = 1;
-    sbr->ready_for_dequant = 0;

    // Save last spectrum parameters variables to compare to new ones
    memcpy(&old_spectrum_params, &sbr->spectrum_params, sizeof(SpectrumParameters));
@@ -621,26 +619,24 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
    int abs_bord_trail = 16;
    int num_rel_lead, num_rel_trail;
    unsigned bs_num_env_old = ch_data->bs_num_env;
-    int bs_frame_class, bs_num_env;

    ch_data->bs_freq_res[0] = ch_data->bs_freq_res[ch_data->bs_num_env];
    ch_data->bs_amp_res = sbr->bs_amp_res_header;
    ch_data->t_env_num_env_old = ch_data->t_env[bs_num_env_old];

-    switch (bs_frame_class = get_bits(gb, 2)) {
+    switch (ch_data->bs_frame_class = get_bits(gb, 2)) {
    case FIXFIX:
-        bs_num_env = 1 << get_bits(gb, 2);
-        if (bs_num_env > 4) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "Invalid bitstream, too many SBR envelopes in FIXFIX type SBR frame: %d\n",
-                   bs_num_env);
-            return -1;
-        }
-        ch_data->bs_num_env = bs_num_env;
+        ch_data->bs_num_env                 = 1 << get_bits(gb, 2);
        num_rel_lead                        = ch_data->bs_num_env - 1;
        if (ch_data->bs_num_env == 1)
            ch_data->bs_amp_res = 0;

+        if (ch_data->bs_num_env > 4) {
+            av_log(ac->avctx, AV_LOG_ERROR,
+                   "Invalid bitstream, too many SBR envelopes in FIXFIX type SBR frame: %d\n",
+                   ch_data->bs_num_env);
+            return -1;
+        }

        ch_data->t_env[0]                   = 0;
        ch_data->t_env[ch_data->bs_num_env] = abs_bord_trail;
@@ -688,15 +684,14 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
        abs_bord_trail                     += get_bits(gb, 2);
        num_rel_lead                        = get_bits(gb, 2);
        num_rel_trail                       = get_bits(gb, 2);
-        bs_num_env                          = num_rel_lead + num_rel_trail + 1;
+        ch_data->bs_num_env                 = num_rel_lead + num_rel_trail + 1;

-        if (bs_num_env > 5) {
+        if (ch_data->bs_num_env > 5) {
            av_log(ac->avctx, AV_LOG_ERROR,
                   "Invalid bitstream, too many SBR envelopes in VARVAR type SBR frame: %d\n",
-                   bs_num_env);
+                   ch_data->bs_num_env);
            return -1;
        }
-        ch_data->bs_num_env = bs_num_env;

        ch_data->t_env[ch_data->bs_num_env] = abs_bord_trail;

@@ -711,7 +706,6 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
        get_bits1_vector(gb, ch_data->bs_freq_res + 1, ch_data->bs_num_env);
        break;
    }
-    ch_data->bs_frame_class = bs_frame_class;

    av_assert0(bs_pointer >= 0);
    if (bs_pointer > ch_data->bs_num_env + 1) {
@@ -722,8 +716,8 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
    }

    for (i = 1; i <= ch_data->bs_num_env; i++) {
-        if (ch_data->t_env[i-1] >= ch_data->t_env[i]) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Not strictly monotone time borders\n");
+        if (ch_data->t_env[i-1] > ch_data->t_env[i]) {
+            av_log(ac->avctx, AV_LOG_ERROR, "Non monotone time borders\n");
            return -1;
        }
    }
@@ -1038,7 +1032,6 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
    unsigned int cnt = get_bits_count(gb);

    sbr->id_aac = id_aac;
-    sbr->ready_for_dequant = 1;

    if (id_aac == TYPE_SCE || id_aac == TYPE_CCE) {
        if (read_sbr_single_channel_element(ac, sbr, gb)) {
@@ -1158,9 +1151,6 @@ static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
                             INTFLOAT z[320], INTFLOAT W[2][32][32][2], int buf_idx)
 {
    int i;
-#if USE_FIXED
-    int j;
-#endif
    memcpy(x    , x+1024, (320-32)*sizeof(x[0]));
    memcpy(x+288, in,         1024*sizeof(x[0]));
    for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
@@ -1168,21 +1158,6 @@ static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
        dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
        sbrdsp->sum64x5(z);
        sbrdsp->qmf_pre_shuffle(z);
-#if USE_FIXED
-        for (j = 64; j < 128; j++) {
-            if (z[j] > 1<<24) {
-                av_log(NULL, AV_LOG_WARNING,
-                       "sbr_qmf_analysis: value %09d too large, setting to %09d\n",
-                       z[j], 1<<24);
-                z[j] = 1<<24;
-            } else if (z[j] < -(1<<24)) {
-                av_log(NULL, AV_LOG_WARNING,
-                       "sbr_qmf_analysis: value %09d too small, setting to %09d\n",
-                       z[j], -(1<<24));
-                z[j] = -(1<<24);
-            }
-        }
-#endif
        mdct->imdct_half(mdct, z, z+64);
        sbrdsp->qmf_post_shuffle(W[buf_idx][i], z);
        x += 32;
@@ -1437,7 +1412,7 @@ static void sbr_env_estimate(AAC_FLOAT (*e_curr)[48], INTFLOAT X_high[64][40][2]

            for (p = 0; p < sbr->n[ch_data->bs_freq_res[e + 1]]; p++) {
 #if USE_FIXED
-                SoftFloat sum = FLOAT_0;
+                SoftFloat sum = { 0, 0 };
                const SoftFloat den = av_int2sf(0x20000000 / (env_size * (table[p + 1] - table[p])), 29);
                for (k = table[p]; k < table[p + 1]; k++) {
                    sum = av_add_sf(sum, sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb));
@@ -1474,12 +1449,6 @@ void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int
        sbr_turnoff(sbr);
    }

-    if (sbr->start && !sbr->ready_for_dequant) {
-        av_log(ac->avctx, AV_LOG_ERROR,
-               "No quantized data read for sbr_dequant.\n");
-        sbr_turnoff(sbr);
-    }
-
    if (!sbr->kx_and_m_pushed) {
        sbr->kx[0] = sbr->kx[1];
        sbr->m[0] = sbr->m[1];
@@ -1489,7 +1458,6 @@ void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int

    if (sbr->start) {
        sbr_dequant(sbr, id_aac);
-        sbr->ready_for_dequant = 0;
    }
    for (ch = 0; ch < nch; ch++) {
        /* decode channel */
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -750,31 +750,30 @@ static void ac3_upmix_delay(AC3DecodeContext *s)
 * @param[in] default_band_struct default band structure table
 * @param[out] num_bands number of bands (optionally NULL)
 * @param[out] band_sizes array containing the number of bins in each band (optionally NULL)
- * @param[in,out] band_struct current band structure
 */
 static void decode_band_structure(GetBitContext *gbc, int blk, int eac3,
                                  int ecpl, int start_subband, int end_subband,
                                  const uint8_t *default_band_struct,
-                                  int *num_bands, uint8_t *band_sizes,
-                                  uint8_t *band_struct, int band_struct_size)
+                                  int *num_bands, uint8_t *band_sizes)
 {
    int subbnd, bnd, n_subbands, n_bands=0;
    uint8_t bnd_sz[22];
+    uint8_t coded_band_struct[22];
+    const uint8_t *band_struct;

    n_subbands = end_subband - start_subband;

-    if (!blk)
-        memcpy(band_struct, default_band_struct, band_struct_size);
-
-    av_assert0(band_struct_size >= start_subband + n_subbands);
-
-    band_struct += start_subband + 1;
-
    /* decode band structure from bitstream or use default */
    if (!eac3 || get_bits1(gbc)) {
        for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) {
-            band_struct[subbnd] = get_bits1(gbc);
+            coded_band_struct[subbnd] = get_bits1(gbc);
        }
+        band_struct = coded_band_struct;
+    } else if (!blk) {
+        band_struct = &default_band_struct[start_subband+1];
+    } else {
+        /* no change in band structure */
+        return;
    }

    /* calculate number of bands and band sizes based on band structure.
@@ -901,15 +900,12 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                                  start_subband, end_subband,
                                  ff_eac3_default_spx_band_struct,
                                  &s->num_spx_bands,
-                                  s->spx_band_sizes,
-                                  s->spx_band_struct, sizeof(s->spx_band_struct));
-        }
-    }
-    if (!s->eac3 || !s->spx_in_use) {
-        s->spx_in_use = 0;
-        for (ch = 1; ch <= fbw_channels; ch++) {
-            s->channel_uses_spx[ch] = 0;
-            s->first_spx_coords[ch] = 1;
+                                  s->spx_band_sizes);
+        } else {
+            for (ch = 1; ch <= fbw_channels; ch++) {
+                s->channel_uses_spx[ch] = 0;
+                s->first_spx_coords[ch] = 1;
+            }
        }
    }

@@ -1036,8 +1032,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
            decode_band_structure(gbc, blk, s->eac3, 0, cpl_start_subband,
                                  cpl_end_subband,
                                  ff_eac3_default_cpl_band_struct,
-                                  &s->num_cpl_bands, s->cpl_band_sizes,
-                                  s->cpl_band_struct, sizeof(s->cpl_band_struct));
+                                  &s->num_cpl_bands, s->cpl_band_sizes);
        } else {
            /* coupling not in use */
            for (ch = 1; ch <= fbw_channels; ch++) {
@@ -1338,7 +1333,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
    for (ch = 1; ch <= s->channels; ch++) {
        int audio_channel = 0;
        INTFLOAT gain;
-        if (s->channel_mode == AC3_CHMODE_DUALMONO && ch <= 2)
+        if (s->channel_mode == AC3_CHMODE_DUALMONO)
            audio_channel = 2-ch;
        if (s->heavy_compression && s->compression_exists[audio_channel])
            gain = s->heavy_dynamic_range[audio_channel];
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -126,7 +126,6 @@ typedef struct AC3DecodeContext {
    int phase_flags_in_use;                 ///< phase flags in use                     (phsflginu)
    int phase_flags[AC3_MAX_CPL_BANDS];     ///< phase flags                            (phsflg)
    int num_cpl_bands;                      ///< number of coupling bands               (ncplbnd)
-    uint8_t cpl_band_struct[AC3_MAX_CPL_BANDS];
    uint8_t cpl_band_sizes[AC3_MAX_CPL_BANDS]; ///< number of coeffs in each coupling band
    int firstchincpl;                       ///< first channel in coupling
    int first_cpl_coords[AC3_MAX_CHANNELS]; ///< first coupling coordinates states      (firstcplcos)
@@ -143,7 +142,6 @@ typedef struct AC3DecodeContext {
    int spx_dst_start_freq;                     ///< spx starting frequency bin for copying (copystartmant)
                                                ///< the copy region ends at the start of the spx region.
    int num_spx_bands;                          ///< number of spx bands                    (nspxbnds)
-    uint8_t spx_band_struct[SPX_MAX_BANDS];
    uint8_t spx_band_sizes[SPX_MAX_BANDS];      ///< number of bins in each spx band
    uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states           (firstspxcos)
    INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor  (nblendfact)
--- a/libavcodec/ac3dec_fixed.c
+++ b/libavcodec/ac3dec_fixed.c
@@ -64,12 +64,12 @@ static void scale_coefs (
    int dynrng,
    int len)
 {
-    int i, shift;
-    unsigned mul, round;
+    int i, shift, round;
+    int16_t mul;
    int temp, temp1, temp2, temp3, temp4, temp5, temp6, temp7;

    mul = (dynrng & 0x1f) + 0x20;
-    shift = 4 - (sign_extend(dynrng, 9) >> 5);
+    shift = 4 - ((dynrng << 23) >> 28);
    if (shift > 0 ) {
      round = 1 << (shift-1);
      for (i=0; i<len; i+=8) {
@@ -107,30 +107,29 @@ static void scale_coefs (
      }
    } else {
      shift = -shift;
-      mul <<= shift;
      for (i=0; i<len; i+=8) {

          temp = src[i] * mul;
          temp1 = src[i+1] * mul;
          temp2 = src[i+2] * mul;

-          dst[i] = temp;
+          dst[i] = temp << shift;
          temp3 = src[i+3] * mul;

-          dst[i+1] = temp1;
+          dst[i+1] = temp1 << shift;
          temp4 = src[i + 4] * mul;
-          dst[i+2] = temp2;
+          dst[i+2] = temp2 << shift;

          temp5 = src[i+5] * mul;
-          dst[i+3] = temp3;
+          dst[i+3] = temp3 << shift;
          temp6 = src[i+6] * mul;

-          dst[i+4] = temp4;
+          dst[i+4] = temp4 << shift;
          temp7 = src[i+7] * mul;

-          dst[i+5] = temp5;
-          dst[i+6] = temp6;
-          dst[i+7] = temp7;
+          dst[i+5] = temp5 << shift;
+          dst[i+6] = temp6 << shift;
+          dst[i+7] = temp7 << shift;

      }
    }
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -2051,8 +2051,7 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
        av_freep(&block->cpl_coord_mant);
    }

-    if (s->mdct_end)
-        s->mdct_end(s);
+    s->mdct_end(s);

    return 0;
 }
@@ -2434,7 +2433,7 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)

    ret = validate_options(s);
    if (ret)
-        goto init_fail;
+        return ret;

    avctx->frame_size = AC3_BLOCK_SIZE * s->num_blocks;
    avctx->initial_padding = AC3_BLOCK_SIZE;
--- a/libavcodec/acelp_pitch_delay.c
+++ b/libavcodec/acelp_pitch_delay.c
@@ -135,7 +135,7 @@ float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
        exp2f(M_LOG2_10 * 0.05 *
              (avpriv_scalarproduct_float_c(pred_table, prediction_error, 4) +
               energy_mean)) /
-        sqrtf(fixed_mean_energy ? fixed_mean_energy : 1.0);
+        sqrtf(fixed_mean_energy);

    // update quantified prediction error energy history
    memmove(&prediction_error[0], &prediction_error[1],
--- a/libavcodec/adpcm.c
+++ b/libavcodec/adpcm.c
@@ -127,8 +127,8 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
        break;
    case AV_CODEC_ID_ADPCM_IMA_APC:
        if (avctx->extradata && avctx->extradata_size >= 8) {
-            c->status[0].predictor = av_clip_intp2(AV_RL32(avctx->extradata    ), 18);
-            c->status[1].predictor = av_clip_intp2(AV_RL32(avctx->extradata + 4), 18);
+            c->status[0].predictor = AV_RL32(avctx->extradata);
+            c->status[1].predictor = AV_RL32(avctx->extradata + 4);
        }
        break;
    case AV_CODEC_ID_ADPCM_IMA_WS:
@@ -277,7 +277,7 @@ static inline short adpcm_ima_oki_expand_nibble(ADPCMChannelStatus *c, int nibbl
    c->predictor = av_clip_intp2(predictor, 11);
    c->step_index = step_index;

-    return c->predictor * 16;
+    return c->predictor << 4;
 }

 static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
@@ -357,10 +357,6 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
            avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
            filter=0;
        }
-        if (shift < 0) {
-            avpriv_request_sample(avctx, "unknown XA-ADPCM shift %d", shift);
-            shift = 0;
-        }
        f0 = xa_adpcm_table[filter][0];
        f1 = xa_adpcm_table[filter][1];

@@ -371,7 +367,7 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
            d = in[16+i+j*4];

            t = sign_extend(d, 4);
-            s = t*(1<<shift) + ((s_1*f0 + s_2*f1+32)>>6);
+            s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
            s_2 = s_1;
            s_1 = av_clip_int16(s);
            out0[j] = s_1;
@@ -386,14 +382,10 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,

        shift  = 12 - (in[5+i*2] & 15);
        filter = in[5+i*2] >> 4;
-        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table) || shift < 0) {
+        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table)) {
            avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
            filter=0;
        }
-        if (shift < 0) {
-            avpriv_request_sample(avctx, "unknown XA-ADPCM shift %d", shift);
-            shift = 0;
-        }

        f0 = xa_adpcm_table[filter][0];
        f1 = xa_adpcm_table[filter][1];
@@ -402,7 +394,7 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
            d = in[16+i+j*4];

            t = sign_extend(d >> 4, 4);
-            s = t*(1<<shift) + ((s_1*f0 + s_2*f1+32)>>6);
+            s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
            s_2 = s_1;
            s_1 = av_clip_int16(s);
            out1[j] = s_1;
@@ -1072,11 +1064,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                return AVERROR_INVALIDDATA;
            }
        }
-        for (i=0; i<=st; i++) {
+        for (i=0; i<=st; i++)
            c->status[i].predictor  = bytestream2_get_le32u(&gb);
-            if (FFABS((int64_t)c->status[i].predictor) > (1<<16))
-                return AVERROR_INVALIDDATA;
-        }

        for (n = nb_samples >> (1 - st); n > 0; n--) {
            int byte   = bytestream2_get_byteu(&gb);
@@ -1224,10 +1213,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,

                    for (count2=0; count2<28; count2++) {
                        if (count2 & 1)
-                            next_sample = (unsigned)sign_extend(byte,    4) << shift;
+                            next_sample = sign_extend(byte,    4) << shift;
                        else {
                            byte = bytestream2_get_byte(&gb);
-                            next_sample = (unsigned)sign_extend(byte >> 4, 4) << shift;
+                            next_sample = sign_extend(byte >> 4, 4) << shift;
                        }

                        next_sample += (current_sample  * coeff1) +
@@ -1487,8 +1476,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                int byte = bytestream2_get_byteu(&gb);
                int index = (byte >> 4) & 7;
                unsigned int exp = byte & 0x0F;
-                int64_t factor1 = table[ch][index * 2];
-                int64_t factor2 = table[ch][index * 2 + 1];
+                int factor1 = table[ch][index * 2];
+                int factor2 = table[ch][index * 2 + 1];

                /* Decode 14 samples.  */
                for (n = 0; n < 14 && (i * 14 + n < nb_samples); n++) {
@@ -1549,7 +1538,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                    else
                        sampledat = sign_extend(byte >> 4, 4);

-                    sampledat = ((sampledat * (1 << 12)) >> (header & 0xf)) * (1 << 6) + prev;
+                    sampledat = (((sampledat << 12) >> (header & 0xf)) << 6) + prev;
                    *samples++ = av_clip_int16(sampledat >> 6);
                    c->status[channel].sample2 = c->status[channel].sample1;
                    c->status[channel].sample1 = sampledat;
--- a/libavcodec/adxdec.c
+++ b/libavcodec/adxdec.c
@@ -81,7 +81,7 @@ static int adx_decode(ADXContext *c, int16_t *out, int offset,
    s2 = prev->s2;
    for (i = 0; i < BLOCK_SAMPLES; i++) {
        d  = get_sbits(&gb, 4);
-        s0 = ((d * (1 << COEFF_BITS)) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
+        s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
        s2 = s1;
        s1 = av_clip_int16(s0);
        *out++ = s1;
--- a/libavcodec/aic.c
+++ b/libavcodec/aic.c
@@ -207,9 +207,6 @@ static int aic_decode_coeffs(GetBitContext *gb, int16_t *dst,
    int mb, idx;
    unsigned val;

-    if (get_bits_left(gb) < 5)
-        return AVERROR_INVALIDDATA;
-
    has_skips  = get_bits1(gb);
    coeff_type = get_bits1(gb);
    coeff_bits = get_bits(gb, 3);
--- a/libavcodec/alac.c
+++ b/libavcodec/alac.c
@@ -167,12 +167,12 @@ static inline int sign_only(int v)
    return v ? FFSIGN(v) : 0;
 }

-static void lpc_prediction(int32_t *error_buffer, uint32_t *buffer_out,
+static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
                           int nb_samples, int bps, int16_t *lpc_coefs,
                           int lpc_order, int lpc_quant)
 {
    int i;
-    uint32_t *pred = buffer_out;
+    int32_t *pred = buffer_out;

    /* first sample always copies */
    *buffer_out = *error_buffer;
@@ -204,27 +204,27 @@ static void lpc_prediction(int32_t *error_buffer, uint32_t *buffer_out,
    for (; i < nb_samples; i++) {
        int j;
        int val = 0;
-        unsigned error_val = error_buffer[i];
+        int error_val = error_buffer[i];
        int error_sign;
        int d = *pred++;

        /* LPC prediction */
        for (j = 0; j < lpc_order; j++)
            val += (pred[j] - d) * lpc_coefs[j];
-        val = (val + (1LL << (lpc_quant - 1))) >> lpc_quant;
+        val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
        val += d + error_val;
        buffer_out[i] = sign_extend(val, bps);

        /* adapt LPC coefficients */
        error_sign = sign_only(error_val);
        if (error_sign) {
-            for (j = 0; j < lpc_order && (int)(error_val * error_sign) > 0; j++) {
+            for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
                int sign;
                val  = d - pred[j];
                sign = sign_only(val) * error_sign;
                lpc_coefs[j] -= sign;
-                val *= (unsigned)sign;
-                error_val -= (val >> lpc_quant) * (j + 1U);
+                val *= sign;
+                error_val -= (val >> lpc_quant) * (j + 1);
            }
        }
    }
@@ -325,16 +325,13 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
        decorr_shift       = get_bits(&alac->gb, 8);
        decorr_left_weight = get_bits(&alac->gb, 8);

-        if (channels == 2 && decorr_left_weight && decorr_shift > 31)
-            return AVERROR_INVALIDDATA;
-
        for (ch = 0; ch < channels; ch++) {
            prediction_type[ch]   = get_bits(&alac->gb, 4);
            lpc_quant[ch]         = get_bits(&alac->gb, 4);
            rice_history_mult[ch] = get_bits(&alac->gb, 3);
            lpc_order[ch]         = get_bits(&alac->gb, 5);

-            if (lpc_order[ch] >= alac->max_samples_per_frame || !lpc_quant[ch])
+            if (lpc_order[ch] >= alac->max_samples_per_frame)
                return AVERROR_INVALIDDATA;

            /* read the predictor table */
--- a/libavcodec/aliaspixdec.c
+++ b/libavcodec/aliaspixdec.c
@@ -62,9 +62,6 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
    if (ret < 0)
        return ret;

-    if (bytestream2_get_bytes_left(&gb) < width*height / 255)
-        return AVERROR_INVALIDDATA;
-
    ret = ff_get_buffer(avctx, f, 0);
    if (ret < 0)
        return ret;
--- a/libavcodec/alsdec.c
+++ b/libavcodec/alsdec.c
@@ -337,11 +337,6 @@ static av_cold int read_specific_config(ALSDecContext *ctx)
    if (als_id != MKBETAG('A','L','S','\0'))
        return AVERROR_INVALIDDATA;

-    if (avctx->channels > FF_SANE_NB_CHANNELS) {
-        avpriv_request_sample(avctx, "Huge number of channels\n");
-        return AVERROR_PATCHWELCOME;
-    }
-
    ctx->cur_frame_length = sconf->frame_length;

    // read channel config
@@ -482,7 +477,7 @@ static void parse_bs_info(const uint32_t bs_info, unsigned int n,
 static int32_t decode_rice(GetBitContext *gb, unsigned int k)
 {
    int max = get_bits_left(gb) - k;
-    unsigned q = get_unary(gb, 0, max);
+    int q   = get_unary(gb, 0, max);
    int r   = k ? get_bits1(gb) : !(q & 1);

    if (k > 1) {
@@ -502,7 +497,7 @@ static void parcor_to_lpc(unsigned int k, const int32_t *par, int32_t *cof)
    int i, j;

    for (i = 0, j = k - 1; i < j; i++, j--) {
-        unsigned tmp1 = ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20);
+        int tmp1 = ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20);
        cof[j]  += ((MUL64(par[k], cof[i]) + (1 << 19)) >> 20);
        cof[i]  += tmp1;
    }
@@ -652,7 +647,7 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)

    // do not continue in case of a damaged stream since
    // block_length must be evenly divisible by sub_blocks
-    if (bd->block_length & (sub_blocks - 1) || bd->block_length <= 0) {
+    if (bd->block_length & (sub_blocks - 1)) {
        av_log(avctx, AV_LOG_WARNING,
               "Block length is not evenly divisible by the number of subblocks.\n");
        return AVERROR_INVALIDDATA;
@@ -767,20 +762,15 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
        if (*bd->use_ltp) {
            int r, c;

-            bd->ltp_gain[0]   = decode_rice(gb, 1) * 8;
-            bd->ltp_gain[1]   = decode_rice(gb, 2) * 8;
+            bd->ltp_gain[0]   = decode_rice(gb, 1) << 3;
+            bd->ltp_gain[1]   = decode_rice(gb, 2) << 3;

-            r                 = get_unary(gb, 0, 4);
+            r                 = get_unary(gb, 0, 3);
            c                 = get_bits(gb, 2);
-            if (r >= 4) {
-                av_log(avctx, AV_LOG_ERROR, "r overflow\n");
-                return AVERROR_INVALIDDATA;
-            }
-
            bd->ltp_gain[2]   = ltp_gain_values[r][c];

-            bd->ltp_gain[3]   = decode_rice(gb, 2) * 8;
-            bd->ltp_gain[4]   = decode_rice(gb, 1) * 8;
+            bd->ltp_gain[3]   = decode_rice(gb, 2) << 3;
+            bd->ltp_gain[4]   = decode_rice(gb, 1) << 3;

            *bd->ltp_lag      = get_bits(gb, ctx->ltp_lag_length);
            *bd->ltp_lag     += FFMAX(4, opt_order + 1);
@@ -789,20 +779,14 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)

    // read first value and residuals in case of a random access block
    if (bd->ra_block) {
-        start = FFMIN(opt_order, 3);
-        av_assert0(sb_length <= sconf->frame_length);
-        if (sb_length <= start) {
-            // opt_order or sb_length may be corrupted, either way this is unsupported and not well defined in the specification
-            av_log(avctx, AV_LOG_ERROR, "Sub block length smaller or equal start\n");
-            return AVERROR_PATCHWELCOME;
-        }
-
        if (opt_order)
            bd->raw_samples[0] = decode_rice(gb, avctx->bits_per_raw_sample - 4);
        if (opt_order > 1)
            bd->raw_samples[1] = decode_rice(gb, FFMIN(s[0] + 3, ctx->s_max));
        if (opt_order > 2)
            bd->raw_samples[2] = decode_rice(gb, FFMIN(s[0] + 1, ctx->s_max));
+
+        start = FFMIN(opt_order, 3);
    }

    // read all residuals
@@ -816,9 +800,7 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
        unsigned int low;
        unsigned int value;

-        int ret = ff_bgmc_decode_init(gb, &high, &low, &value);
-        if (ret < 0)
-            return ret;
+        ff_bgmc_decode_init(gb, &high, &low, &value);

        current_res = bd->raw_samples + start;

@@ -828,9 +810,6 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
            k    [sb] = s[sb] > b ? s[sb] - b : 0;
            delta[sb] = 5 - s[sb] + k[sb];

-            if (k[sb] >= 32)
-                return AVERROR_INVALIDDATA;
-
            ff_bgmc_decode(gb, sb_len, current_res,
                        delta[sb], sx[sb], &high, &low, &value, ctx->bgmc_lut, ctx->bgmc_lut_status);

@@ -926,7 +905,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
            y = 1 << 6;

            for (base = begin; base < end; base++, tab++)
-                y += (uint64_t)MUL64(bd->ltp_gain[tab], raw_samples[base]);
+                y += MUL64(bd->ltp_gain[tab], raw_samples[base]);

            raw_samples[ltp_smp] += y >> 7;
        }
@@ -938,7 +917,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
            y = 1 << 19;

            for (sb = 0; sb < smp; sb++)
-                y += (uint64_t)MUL64(lpc_cof[sb], raw_samples[-(sb + 1)]);
+                y += MUL64(lpc_cof[sb], raw_samples[-(sb + 1)]);

            *raw_samples++ -= y >> 20;
            parcor_to_lpc(smp, quant_cof, lpc_cof);
@@ -954,7 +933,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)

        // reconstruct difference signal for prediction (joint-stereo)
        if (bd->js_blocks && bd->raw_other) {
-            uint32_t *left, *right;
+            int32_t *left, *right;

            if (bd->raw_other > raw_samples) {  // D = R - L
                left  = raw_samples;
@@ -988,7 +967,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
        y = 1 << 19;

        for (sb = -opt_order; sb < 0; sb++)
-            y += (uint64_t)MUL64(lpc_cof[sb], raw_samples[sb]);
+            y += MUL64(lpc_cof[sb], raw_samples[sb]);

        *raw_samples -= y >> 20;
    }
@@ -1012,10 +991,6 @@ static int read_block(ALSDecContext *ctx, ALSBlockData *bd)
    GetBitContext *gb        = &ctx->gb;

    *bd->shift_lsbs = 0;
-
-    if (get_bits_left(gb) < 7)
-        return AVERROR_INVALIDDATA;
-
    // read block type flag and read the samples accordingly
    if (get_bits1(gb)) {
        ret = read_var_block_data(ctx, bd);
@@ -1047,7 +1022,7 @@ static int decode_block(ALSDecContext *ctx, ALSBlockData *bd)

    if (*bd->shift_lsbs)
        for (smp = 0; smp < bd->block_length; smp++)
-            bd->raw_samples[smp] = (unsigned)bd->raw_samples[smp] << *bd->shift_lsbs;
+            bd->raw_samples[smp] <<= *bd->shift_lsbs;

    return 0;
 }
@@ -1183,10 +1158,10 @@ static int decode_blocks(ALSDecContext *ctx, unsigned int ra_frame,
                av_log(ctx->avctx, AV_LOG_WARNING, "Invalid channel pair.\n");

            for (s = 0; s < div_blocks[b]; s++)
-                bd[0].raw_samples[s] = bd[1].raw_samples[s] - (unsigned)bd[0].raw_samples[s];
+                bd[0].raw_samples[s] = bd[1].raw_samples[s] - bd[0].raw_samples[s];
        } else if (bd[1].js_blocks) {
            for (s = 0; s < div_blocks[b]; s++)
-                bd[1].raw_samples[s] = bd[1].raw_samples[s] + (unsigned)bd[0].raw_samples[s];
+                bd[1].raw_samples[s] = bd[1].raw_samples[s] + bd[0].raw_samples[s];
        }

        offset  += div_blocks[b];
@@ -1572,17 +1547,15 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
    #define INTERLEAVE_OUTPUT(bps)                                                   \
    {                                                                                \
        int##bps##_t *dest = (int##bps##_t*)frame->data[0];                          \
-        int channels = avctx->channels;                                              \
-        int32_t **raw_samples = ctx->raw_samples;                                    \
        shift = bps - ctx->avctx->bits_per_raw_sample;                               \
        if (!ctx->cs_switch) {                                                       \
            for (sample = 0; sample < ctx->cur_frame_length; sample++)               \
-                for (c = 0; c < channels; c++)                                       \
-                    *dest++ = raw_samples[c][sample] * (1U << shift);                \
+                for (c = 0; c < avctx->channels; c++)                                \
+                    *dest++ = ctx->raw_samples[c][sample] << shift;                  \
        } else {                                                                     \
            for (sample = 0; sample < ctx->cur_frame_length; sample++)               \
-                for (c = 0; c < channels; c++)                                       \
-                    *dest++ = raw_samples[sconf->chan_pos[c]][sample] * (1U << shift);\
+                for (c = 0; c < avctx->channels; c++)                                \
+                    *dest++ = ctx->raw_samples[sconf->chan_pos[c]][sample] << shift; \
        }                                                                            \
    }

@@ -1749,8 +1722,6 @@ static av_cold int decode_init(AVCodecContext *avctx)

    // allocate quantized parcor coefficient buffer
    num_buffers = sconf->mc_coding ? avctx->channels : 1;
-    if (num_buffers * (uint64_t)num_buffers > INT_MAX) // protect chan_data_buffer allocation
-        return AVERROR_INVALIDDATA;

    ctx->quant_cof        = av_malloc_array(num_buffers, sizeof(*ctx->quant_cof));
    ctx->lpc_cof          = av_malloc_array(num_buffers, sizeof(*ctx->lpc_cof));
@@ -1857,6 +1828,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
    return 0;

 fail:
+    decode_end(avctx);
    return ret;
 }

@@ -1882,5 +1854,4 @@ AVCodec ff_als_decoder = {
    .decode         = decode_frame,
    .flush          = flush,
    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
-    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
 };
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -265,7 +265,7 @@ static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
            *lag_frac = pitch_index - (*lag_int << 2) + 136;
        } else if (pitch_index < 440) {
            *lag_int  = (pitch_index + 257 - 376) >> 1;
-            *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
+            *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
            /* the actual resolution is 1/2 but expressed as 1/4 */
        } else {
            *lag_int  = pitch_index - 280;
@@ -295,7 +295,7 @@ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
    if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
        if (pitch_index < 116) {
            *lag_int  = (pitch_index + 69) >> 1;
-            *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
+            *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
        } else {
            *lag_int  = pitch_index - 24;
            *lag_frac = 0;
@@ -305,7 +305,7 @@ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
    } else {
        *lag_int  = (pitch_index + 1) >> 1;
-        *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
+        *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
        *lag_int += *base_lag_int;
    }
 }
@@ -614,7 +614,7 @@ static float voice_factor(float *p_vector, float p_gain,
                                                          AMRWB_SFR_SIZE) *
                    f_gain * f_gain;

-    return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
+    return (p_ener - f_ener) / (p_ener + f_ener);
 }

 /**
@@ -865,20 +865,15 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
 {
    int wsp = (vad > 0);
    float tilt;
-    float tmp;

    if (ctx->fr_cur_mode == MODE_23k85)
        return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));

-    tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
-
-    if (tmp > 0) {
-        tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
-    } else
-        tilt = 0;
+    tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
+           ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);

    /* return gain bounded by [0.1, 1.0] */
-    return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
+    return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
 }

 /**
--- a/libavcodec/anm.c
+++ b/libavcodec/anm.c
@@ -119,9 +119,6 @@ static int decode_frame(AVCodecContext *avctx,
    uint8_t *dst, *dst_end;
    int count, ret;

-    if (buf_size < 7)
-        return AVERROR_INVALIDDATA;
-
    if ((ret = ff_reget_buffer(avctx, s->frame)) < 0)
        return ret;
    dst     = s->frame->data[0];
--- a/libavcodec/ansi.c
+++ b/libavcodec/ansi.c
@@ -80,6 +80,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
    AnsiContext *s = avctx->priv_data;
    avctx->pix_fmt = AV_PIX_FMT_PAL8;

+    s->frame = av_frame_alloc();
+    if (!s->frame)
+        return AVERROR(ENOMEM);
+
    /* defaults */
    s->font        = avpriv_vga16_font;
    s->font_height = 16;
@@ -90,15 +94,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
        int ret = ff_set_dimensions(avctx, 80 << 3, 25 << 4);
        if (ret < 0)
            return ret;
-    } else if (avctx->width % FONT_WIDTH || avctx->height % s->font_height) {
-        av_log(avctx, AV_LOG_ERROR, "Invalid dimensions %d %d\n", avctx->width, avctx->height);
-        return AVERROR(EINVAL);
    }
-
-    s->frame = av_frame_alloc();
-    if (!s->frame)
-        return AVERROR(ENOMEM);
-
    return 0;
 }

@@ -430,8 +426,7 @@ static int decode_frame(AVCodecContext *avctx,
                    s->args[s->nb_args] = FFMAX(s->args[s->nb_args], 0) * 10 + buf[0] - '0';
                break;
            case ';':
-                if (s->nb_args < MAX_NB_ARGS)
-                    s->nb_args++;
+                s->nb_args++;
                if (s->nb_args < MAX_NB_ARGS)
                    s->args[s->nb_args] = 0;
                break;
--- a/libavcodec/apedec.c
+++ b/libavcodec/apedec.c
@@ -101,7 +101,7 @@ typedef struct APEFilter {
    int16_t *historybuffer; ///< filter memory
    int16_t *delay;         ///< filtered values

-    uint32_t avg;
+    int avg;
 } APEFilter;

 typedef struct APERice {
@@ -460,7 +460,7 @@ static inline void update_rice(APERice *rice, unsigned int x)

    if (rice->ksum < lim)
        rice->k--;
-    else if (rice->ksum >= (1 << (rice->k + 5)) && rice->k < 24)
+    else if (rice->ksum >= (1 << (rice->k + 5)))
        rice->k++;
 }

@@ -554,7 +554,7 @@ static inline int ape_decode_value_3990(APEContext *ctx, APERice *rice)
    overflow = range_get_symbol(ctx, counts_3980, counts_diff_3980);

    if (overflow == (MODEL_ELEMENTS - 1)) {
-        overflow  = (unsigned)range_decode_bits(ctx, 16) << 16;
+        overflow  = range_decode_bits(ctx, 16) << 16;
        overflow |= range_decode_bits(ctx, 16);
    }

@@ -589,7 +589,7 @@ static void decode_array_0000(APEContext *ctx, GetBitContext *gb,
                              int32_t *out, APERice *rice, int blockstodecode)
 {
    int i;
-    unsigned ksummax, ksummin;
+    int ksummax, ksummin;

    rice->ksum = 0;
    for (i = 0; i < FFMIN(blockstodecode, 5); i++) {
@@ -610,7 +610,7 @@ static void decode_array_0000(APEContext *ctx, GetBitContext *gb,
    ksummin = rice->k ? (1 << rice->k + 6) : 0;
    for (; i < blockstodecode; i++) {
        out[i] = get_rice_ook(&ctx->gb, rice->k);
-        rice->ksum += out[i] - (unsigned)out[i - 64];
+        rice->ksum += out[i] - out[i - 64];
        while (rice->ksum < ksummin) {
            rice->k--;
            ksummin = rice->k ? ksummin >> 1 : 0;
@@ -836,7 +836,7 @@ static av_always_inline int filter_fast_3320(APEPredictor *p,
    else
        p->coeffsA[filter][0]--;

-    p->filterA[filter] += (unsigned)p->lastA[filter];
+    p->filterA[filter] += p->lastA[filter];

    return p->filterA[filter];
 }
@@ -859,9 +859,9 @@ static av_always_inline int filter_3800(APEPredictor *p,
        return predictionA;
    }
    d2 =  p->buf[delayA];
-    d1 = (p->buf[delayA] - (unsigned)p->buf[delayA - 1]) * 2;
-    d0 =  p->buf[delayA] + ((p->buf[delayA - 2] - (unsigned)p->buf[delayA - 1]) * 8);
-    d3 =  p->buf[delayB] * 2U - p->buf[delayB - 1];
+    d1 = (p->buf[delayA] - p->buf[delayA - 1]) << 1;
+    d0 =  p->buf[delayA] + ((p->buf[delayA - 2] - p->buf[delayA - 1]) << 3);
+    d3 =  p->buf[delayB] * 2 - p->buf[delayB - 1];
    d4 =  p->buf[delayB];

    predictionA = d0 * p->coeffsA[filter][0] +
@@ -880,8 +880,8 @@ static av_always_inline int filter_3800(APEPredictor *p,
    p->coeffsB[filter][0] += (((d3 >> 29) & 4) - 2) * sign;
    p->coeffsB[filter][1] -= (((d4 >> 30) & 2) - 1) * sign;

-    p->filterB[filter] = p->lastA[filter] + (unsigned)(predictionB >> shift);
-    p->filterA[filter] = p->filterB[filter] + (unsigned)((int)(p->filterA[filter] * 31U) >> 5);
+    p->filterB[filter] = p->lastA[filter] + (predictionB >> shift);
+    p->filterA[filter] = p->filterB[filter] + ((p->filterA[filter] * 31) >> 5);

    return p->filterA[filter];
 }
@@ -892,9 +892,6 @@ static void long_filter_high_3800(int32_t *buffer, int order, int shift, int len
    int32_t dotprod, sign;
    int32_t coeffs[256], delay[256];

-    if (order >= length)
-        return;
-
    memset(coeffs, 0, order * sizeof(*coeffs));
    for (i = 0; i < order; i++)
        delay[i] = buffer[i];
@@ -902,10 +899,10 @@ static void long_filter_high_3800(int32_t *buffer, int order, int shift, int len
        dotprod = 0;
        sign = APESIGN(buffer[i]);
        for (j = 0; j < order; j++) {
-            dotprod += delay[j] * (unsigned)coeffs[j];
+            dotprod += delay[j] * coeffs[j];
            coeffs[j] += ((delay[j] >> 31) | 1) * sign;
        }
-        buffer[i] -= (unsigned)(dotprod >> shift);
+        buffer[i] -= dotprod >> shift;
        for (j = 0; j < order - 1; j++)
            delay[j] = delay[j + 1];
        delay[order - 1] = buffer[i];
@@ -916,8 +913,7 @@ static void long_filter_ehigh_3830(int32_t *buffer, int length)
 {
    int i, j;
    int32_t dotprod, sign;
-    int32_t delay[8] = { 0 };
-    uint32_t coeffs[8] = { 0 };
+    int32_t coeffs[8] = { 0 }, delay[8] = { 0 };

    for (i = 0; i < length; i++) {
        dotprod = 0;
@@ -929,7 +925,7 @@ static void long_filter_ehigh_3830(int32_t *buffer, int length)
        for (j = 7; j > 0; j--)
            delay[j] = delay[j - 1];
        delay[0] = buffer[i];
-        buffer[i] -= (unsigned)(dotprod >> 9);
+        buffer[i] -= dotprod >> 9;
    }
 }

@@ -1038,13 +1034,13 @@ static av_always_inline int predictor_update_3930(APEPredictor *p,
                                                  const int delayA)
 {
    int32_t predictionA, sign;
-    uint32_t d0, d1, d2, d3;
+    int32_t d0, d1, d2, d3;

    p->buf[delayA]     = p->lastA[filter];
    d0 = p->buf[delayA    ];
-    d1 = p->buf[delayA    ] - (unsigned)p->buf[delayA - 1];
-    d2 = p->buf[delayA - 1] - (unsigned)p->buf[delayA - 2];
-    d3 = p->buf[delayA - 2] - (unsigned)p->buf[delayA - 3];
+    d1 = p->buf[delayA    ] - p->buf[delayA - 1];
+    d2 = p->buf[delayA - 1] - p->buf[delayA - 2];
+    d3 = p->buf[delayA - 2] - p->buf[delayA - 3];

    predictionA = d0 * p->coeffsA[filter][0] +
                  d1 * p->coeffsA[filter][1] +
@@ -1052,13 +1048,13 @@ static av_always_inline int predictor_update_3930(APEPredictor *p,
                  d3 * p->coeffsA[filter][3];

    p->lastA[filter] = decoded + (predictionA >> 9);
-    p->filterA[filter] = p->lastA[filter] + ((int)(p->filterA[filter] * 31U) >> 5);
+    p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);

    sign = APESIGN(decoded);
-    p->coeffsA[filter][0] += (((int32_t)d0 < 0) * 2 - 1) * sign;
-    p->coeffsA[filter][1] += (((int32_t)d1 < 0) * 2 - 1) * sign;
-    p->coeffsA[filter][2] += (((int32_t)d2 < 0) * 2 - 1) * sign;
-    p->coeffsA[filter][3] += (((int32_t)d3 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][0] += ((d0 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][1] += ((d1 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][2] += ((d2 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][3] += ((d3 < 0) * 2 - 1) * sign;

    return p->filterA[filter];
 }
@@ -1122,7 +1118,7 @@ static av_always_inline int predictor_update_filter(APEPredictor *p,

    p->buf[delayA]     = p->lastA[filter];
    p->buf[adaptA]     = APESIGN(p->buf[delayA]);
-    p->buf[delayA - 1] = p->buf[delayA] - (unsigned)p->buf[delayA - 1];
+    p->buf[delayA - 1] = p->buf[delayA] - p->buf[delayA - 1];
    p->buf[adaptA - 1] = APESIGN(p->buf[delayA - 1]);

    predictionA = p->buf[delayA    ] * p->coeffsA[filter][0] +
@@ -1131,9 +1127,9 @@ static av_always_inline int predictor_update_filter(APEPredictor *p,
                  p->buf[delayA - 3] * p->coeffsA[filter][3];

    /*  Apply a scaled first-order filter compression */
-    p->buf[delayB]     = p->filterA[filter ^ 1] - ((int)(p->filterB[filter] * 31U) >> 5);
+    p->buf[delayB]     = p->filterA[filter ^ 1] - ((p->filterB[filter] * 31) >> 5);
    p->buf[adaptB]     = APESIGN(p->buf[delayB]);
-    p->buf[delayB - 1] = p->buf[delayB] - (unsigned)p->buf[delayB - 1];
+    p->buf[delayB - 1] = p->buf[delayB] - p->buf[delayB - 1];
    p->buf[adaptB - 1] = APESIGN(p->buf[delayB - 1]);
    p->filterB[filter] = p->filterA[filter ^ 1];

@@ -1143,8 +1139,8 @@ static av_always_inline int predictor_update_filter(APEPredictor *p,
                  p->buf[delayB - 3] * p->coeffsB[filter][3] +
                  p->buf[delayB - 4] * p->coeffsB[filter][4];

-    p->lastA[filter] = decoded + ((int)((unsigned)predictionA + (predictionB >> 1)) >> 10);
-    p->filterA[filter] = p->lastA[filter] + ((int)(p->filterA[filter] * 31U) >> 5);
+    p->lastA[filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
+    p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);

    sign = APESIGN(decoded);
    p->coeffsA[filter][0] += p->buf[adaptA    ] * sign;
@@ -1203,14 +1199,14 @@ static void predictor_decode_mono_3950(APEContext *ctx, int count)
        A = *decoded0;

        p->buf[YDELAYA] = currentA;
-        p->buf[YDELAYA - 1] = p->buf[YDELAYA] - (unsigned)p->buf[YDELAYA - 1];
+        p->buf[YDELAYA - 1] = p->buf[YDELAYA] - p->buf[YDELAYA - 1];

        predictionA = p->buf[YDELAYA    ] * p->coeffsA[0][0] +
                      p->buf[YDELAYA - 1] * p->coeffsA[0][1] +
                      p->buf[YDELAYA - 2] * p->coeffsA[0][2] +
                      p->buf[YDELAYA - 3] * p->coeffsA[0][3];

-        currentA = A + (unsigned)(predictionA >> 10);
+        currentA = A + (predictionA >> 10);

        p->buf[YADAPTCOEFFSA]     = APESIGN(p->buf[YDELAYA    ]);
        p->buf[YADAPTCOEFFSA - 1] = APESIGN(p->buf[YDELAYA - 1]);
@@ -1230,7 +1226,7 @@ static void predictor_decode_mono_3950(APEContext *ctx, int count)
            p->buf = p->historybuffer;
        }

-        p->filterA[0] = currentA + (unsigned)((int)(p->filterA[0] * 31U) >> 5);
+        p->filterA[0] = currentA + ((p->filterA[0] * 31) >> 5);
        *(decoded0++) = p->filterA[0];
    }

@@ -1267,8 +1263,8 @@ static void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
                                                     f->delay - order,
                                                     f->adaptcoeffs - order,
                                                     order, APESIGN(*data));
-        res = (int)(res + (1U << (fracbits - 1))) >> fracbits;
-        res += (unsigned)*data;
+        res = (res + (1 << (fracbits - 1))) >> fracbits;
+        res += *data;
        *data++ = res;

        /* Update the output history */
@@ -1283,14 +1279,14 @@ static void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
            /* Version 3.98 and later files */

            /* Update the adaption coefficients */
-            absres = res < 0 ? -(unsigned)res : res;
+            absres = FFABS(res);
            if (absres)
                *f->adaptcoeffs = ((res & (-1<<31)) ^ (-1<<30)) >>
                                  (25 + (absres <= f->avg*3) + (absres <= f->avg*4/3));
            else
                *f->adaptcoeffs = 0;

-            f->avg += (int)(absres - (unsigned)f->avg) / 16;
+            f->avg += (absres - f->avg) / 16;

            f->adaptcoeffs[-1] >>= 1;
            f->adaptcoeffs[-2] >>= 1;
@@ -1369,11 +1365,11 @@ static void ape_unpack_mono(APEContext *ctx, int count)

 static void ape_unpack_stereo(APEContext *ctx, int count)
 {
-    unsigned left, right;
+    int32_t left, right;
    int32_t *decoded0 = ctx->decoded[0];
    int32_t *decoded1 = ctx->decoded[1];

-    if ((ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) == APE_FRAMECODE_STEREO_SILENCE) {
+    if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
        /* We are pure silence, so we're done. */
        av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence stereo\n");
        return;
@@ -1386,7 +1382,7 @@ static void ape_unpack_stereo(APEContext *ctx, int count)

    /* Decorrelate and scale to output depth */
    while (count--) {
-        left = *decoded1 - (unsigned)(*decoded0 / 2);
+        left = *decoded1 - (*decoded0 / 2);
        right = left + *decoded0;

        *(decoded0++) = left;
@@ -1405,7 +1401,6 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
    int32_t *sample24;
    int i, ch, ret;
    int blockstodecode;
-    uint64_t decoded_buffer_size;

    /* this should never be negative, but bad things will happen if it is, so
       check it just to make sure. */
@@ -1444,8 +1439,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
        if (s->fileversion >= 3900) {
            if (offset > 3) {
                av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n");
-                av_freep(&s->data);
-                s->data_size = 0;
+                s->data = NULL;
                return AVERROR_INVALIDDATA;
            }
            if (s->data_end - s->ptr < offset) {
@@ -1462,7 +1456,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
                skip_bits_long(&s->gb, offset);
        }

-        if (!nblocks || nblocks > INT_MAX / 2 / sizeof(*s->decoded_buffer) - 8) {
+        if (!nblocks || nblocks > INT_MAX) {
            av_log(avctx, AV_LOG_ERROR, "Invalid sample count: %"PRIu32".\n",
                   nblocks);
            return AVERROR_INVALIDDATA;
@@ -1488,12 +1482,11 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
        blockstodecode = s->samples;

    /* reallocate decoded sample buffer if needed */
-    decoded_buffer_size = 2LL * FFALIGN(blockstodecode, 8) * sizeof(*s->decoded_buffer);
-    av_assert0(decoded_buffer_size <= INT_MAX);
-    av_fast_malloc(&s->decoded_buffer, &s->decoded_size, decoded_buffer_size);
+    av_fast_malloc(&s->decoded_buffer, &s->decoded_size,
+                   2 * FFALIGN(blockstodecode, 8) * sizeof(*s->decoded_buffer));
    if (!s->decoded_buffer)
        return AVERROR(ENOMEM);
-    memset(s->decoded_buffer, 0, decoded_buffer_size);
+    memset(s->decoded_buffer, 0, s->decoded_size);
    s->decoded[0] = s->decoded_buffer;
    s->decoded[1] = s->decoded_buffer + FFALIGN(blockstodecode, 8);

@@ -1521,7 +1514,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
        for (ch = 0; ch < s->channels; ch++) {
            sample8 = (uint8_t *)frame->data[ch];
            for (i = 0; i < blockstodecode; i++)
-                *sample8++ = (s->decoded[ch][i] + 0x80U) & 0xff;
+                *sample8++ = (s->decoded[ch][i] + 0x80) & 0xff;
        }
        break;
    case 16:
@@ -1535,7 +1528,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
        for (ch = 0; ch < s->channels; ch++) {
            sample24 = (int32_t *)frame->data[ch];
            for (i = 0; i < blockstodecode; i++)
-                *sample24++ = s->decoded[ch][i] * 256U;
+                *sample24++ = s->decoded[ch][i] << 8;
        }
        break;
    }
--- a/libavcodec/arm/sbrdsp_neon.S
+++ b/libavcodec/arm/sbrdsp_neon.S
@@ -336,11 +336,11 @@ function ff_sbr_hf_apply_noise_0_neon, export=1
        vld1.32         {d0},     [r0,:64]
        vld1.32         {d6},     [lr,:64]
        vld1.32         {d2[]},   [r1,:32]!
-        vld1.32         {d18[]},  [r2,:32]!
+        vld1.32         {d3[]},   [r2,:32]!
        vceq.f32        d4,  d2,  #0
        veor            d2,  d2,  d3
        vmov            d1,  d0
-        vmla.f32        d0,  d6,  d18
+        vmla.f32        d0,  d6,  d3
        vadd.f32        s2,  s2,  s4
        vbif            d0,  d1,  d4
        vst1.32         {d0},     [r0,:64]!
--- a/libavcodec/ass_split.c
+++ b/libavcodec/ass_split.c
@@ -525,7 +525,7 @@ ASSStyle *ff_ass_style_get(ASSSplitContext *ctx, const char *style)
    if (!style || !*style)
        style = "Default";
    for (i=0; i<ass->styles_count; i++)
-        if (ass->styles[i].name && !strcmp(ass->styles[i].name, style))
+        if (!strcmp(ass->styles[i].name, style))
            return ass->styles + i;
    return NULL;
 }
--- a/libavcodec/assdec.c
+++ b/libavcodec/assdec.c
@@ -32,8 +32,7 @@ static av_cold int ass_decode_init(AVCodecContext *avctx)
    avctx->subtitle_header = av_malloc(avctx->extradata_size + 1);
    if (!avctx->subtitle_header)
        return AVERROR(ENOMEM);
-    if (avctx->extradata_size)
-        memcpy(avctx->subtitle_header, avctx->extradata, avctx->extradata_size);
+    memcpy(avctx->subtitle_header, avctx->extradata, avctx->extradata_size);
    avctx->subtitle_header[avctx->extradata_size] = 0;
    avctx->subtitle_header_size = avctx->extradata_size;
    avctx->priv_data = ff_ass_split(avctx->extradata);
--- a/libavcodec/asvenc.c
+++ b/libavcodec/asvenc.c
@@ -61,7 +61,7 @@ static inline void asv2_put_level(ASV1Context *a, PutBitContext *pb, int level)
    } else {
        put_bits(pb, ff_asv2_level_tab[31][1], ff_asv2_level_tab[31][0]);
        if (level < -128 || level > 127) {
-            av_log(a->avctx, AV_LOG_WARNING, "Clipping level %d, increase qscale\n", level);
+            av_log(a->avctx, AV_LOG_WARNING, "Cliping level %d, increase qscale\n", level);
            level = av_clip_int8(level);
        }
        asv2_put_bits(pb, 8, level & 0xFF);
--- a/libavcodec/atrac3.c
+++ b/libavcodec/atrac3.c
@@ -882,7 +882,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
        return AVERROR_INVALIDDATA;
    }

-    if (avctx->block_align > 1024 || avctx->block_align <= 0)
+    if (avctx->block_align >= UINT_MAX / 2)
        return AVERROR(EINVAL);

    q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
--- a/libavcodec/atrac3plus.c
+++ b/libavcodec/atrac3plus.c
@@ -459,10 +459,6 @@ static int decode_channel_wordlen(GetBitContext *gb, Atrac3pChanUnitCtx *ctx,
    } else if (chan->fill_mode == 3) {
        pos = ch_num ? chan->num_coded_vals + chan->split_point
                     : ctx->num_quant_units - chan->split_point;
-        if (pos > FF_ARRAY_ELEMS(chan->qu_wordlen)) {
-            av_log(avctx, AV_LOG_ERROR, "Split point beyond array\n");
-            pos = FF_ARRAY_ELEMS(chan->qu_wordlen);
-        }
        for (i = chan->num_coded_vals; i < pos; i++)
            chan->qu_wordlen[i] = 1;
    }
--- a/libavcodec/audiodsp.c
+++ b/libavcodec/audiodsp.c
@@ -79,7 +79,7 @@ static void vector_clipf_c(float *dst, const float *src,
 static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2,
                                     int order)
 {
-    unsigned res = 0;
+    int res = 0;

    while (order--)
        res += *v1++ **v2++;
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -1365,16 +1365,6 @@ enum AVPacketSideDataType {
     * side data includes updated metadata which appeared in the stream.
     */
    AV_PKT_DATA_METADATA_UPDATE,
-
-    /**
-     * The number of side data elements (in fact a bit more than it).
-     * This is not part of the public API/ABI in the sense that it may
-     * change when new side data types are added.
-     * This must stay the last enum value.
-     * If its value becomes huge, some code using it
-     * needs to be updated as it assumes it to be smaller than other limits.
-     */
-    AV_PKT_DATA_NB
 };

 #define AV_PKT_DATA_QUALITY_FACTOR AV_PKT_DATA_QUALITY_STATS //DEPRECATED
--- a/libavcodec/avdct.c
+++ b/libavcodec/avdct.c
@@ -109,7 +109,7 @@ int avcodec_dct_init(AVDCT *dsp)

 #if CONFIG_IDCTDSP
    {
-        IDCTDSPContext idsp = {0};
+        IDCTDSPContext idsp;
        ff_idctdsp_init(&idsp, avctx);
        COPY(idsp, idct);
        COPY(idsp, idct_permutation);
--- a/libavcodec/avpacket.c
+++ b/libavcodec/avpacket.c
@@ -58,7 +58,6 @@ void av_init_packet(AVPacket *pkt)
 #if FF_API_DESTRUCT_PACKET
 FF_DISABLE_DEPRECATION_WARNINGS
    pkt->destruct             = NULL;
-    pkt->priv                 = NULL;
 FF_ENABLE_DEPRECATION_WARNINGS
 #endif
    pkt->buf                  = NULL;
@@ -195,7 +194,6 @@ static int copy_packet_data(AVPacket *pkt, const AVPacket *src, int dup)
 {
    pkt->data      = NULL;
    pkt->side_data = NULL;
-    pkt->side_data_elems = 0;
    if (pkt->buf) {
        AVBufferRef *ref = av_buffer_ref(src->buf);
        if (!ref)
@@ -210,11 +208,9 @@ FF_DISABLE_DEPRECATION_WARNINGS
    pkt->destruct = dummy_destruct_packet;
 FF_ENABLE_DEPRECATION_WARNINGS
 #endif
-    if (src->side_data_elems && dup) {
+    if (pkt->side_data_elems && dup)
        pkt->side_data = src->side_data;
-        pkt->side_data_elems = src->side_data_elems;
-    }
-    if (src->side_data_elems && !dup) {
+    if (pkt->side_data_elems && !dup) {
        return av_copy_packet_side_data(pkt, src);
    }
    return 0;
@@ -305,12 +301,11 @@ uint8_t *av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
 {
    int elems = pkt->side_data_elems;

-    if ((unsigned)elems + 1 > AV_PKT_DATA_NB)
+    if ((unsigned)elems + 1 > INT_MAX / sizeof(*pkt->side_data))
        return NULL;
    if ((unsigned)size > INT_MAX - AV_INPUT_BUFFER_PADDING_SIZE)
        return NULL;

-
    pkt->side_data = av_realloc(pkt->side_data,
                                (elems + 1) * sizeof(*pkt->side_data));
    if (!pkt->side_data)
@@ -338,8 +333,6 @@ uint8_t *av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
            return pkt->side_data[i].data;
        }
    }
-    if (size)
-        *size = 0;
    return NULL;
 }

@@ -417,18 +410,13 @@ int av_packet_split_side_data(AVPacket *pkt){
        p = pkt->data + pkt->size - 8 - 5;
        for (i=1; ; i++){
            size = AV_RB32(p);
-            if (size>INT_MAX - 5 || p - pkt->data < size)
+            if (size>INT_MAX || p - pkt->data < size)
                return 0;
            if (p[4]&128)
                break;
-            if (p - pkt->data < size + 5)
-                return 0;
            p-= size+5;
        }

-        if (i > AV_PKT_DATA_NB)
-            return AVERROR(ERANGE);
-
        pkt->side_data = av_malloc_array(i, sizeof(*pkt->side_data));
        if (!pkt->side_data)
            return AVERROR(ENOMEM);
@@ -436,7 +424,7 @@ int av_packet_split_side_data(AVPacket *pkt){
        p= pkt->data + pkt->size - 8 - 5;
        for (i=0; ; i++){
            size= AV_RB32(p);
-            av_assert0(size<=INT_MAX - 5 && p - pkt->data >= size);
+            av_assert0(size<=INT_MAX && p - pkt->data >= size);
            pkt->side_data[i].data = av_mallocz(size + AV_INPUT_BUFFER_PADDING_SIZE);
            pkt->side_data[i].size = size;
            pkt->side_data[i].type = p[4]&127;
--- a/libavcodec/avrndec.c
+++ b/libavcodec/avrndec.c
@@ -113,7 +113,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
            int shift = p->height - avctx->height;
            int subsample_h, subsample_v;

-            av_pix_fmt_get_chroma_sub_sample(p->format, &subsample_h, &subsample_v);
+            av_pix_fmt_get_chroma_sub_sample(avctx->pix_fmt, &subsample_h, &subsample_v);

            p->data[0] += p->linesize[0] * shift;
            if (p->data[2]) {
--- a/libavcodec/bethsoftvideo.c
+++ b/libavcodec/bethsoftvideo.c
@@ -109,11 +109,6 @@ static int bethsoftvid_decode_frame(AVCodecContext *avctx,
            if(yoffset >= avctx->height)
                return AVERROR_INVALIDDATA;
            dst += vid->frame->linesize[0] * yoffset;
-        case VIDEO_P_FRAME:
-        case VIDEO_I_FRAME:
-            break;
-        default:
-            return AVERROR_INVALIDDATA;
    }

    // main code
--- a/libavcodec/bgmc.c
+++ b/libavcodec/bgmc.c
@@ -485,17 +485,12 @@ av_cold void ff_bgmc_end(uint8_t **cf_lut, int **cf_lut_status)


 /** Initialize decoding and reads the first value */
-int ff_bgmc_decode_init(GetBitContext *gb, unsigned int *h,
+void ff_bgmc_decode_init(GetBitContext *gb, unsigned int *h,
                         unsigned int *l, unsigned int *v)
 {
-    if (get_bits_left(gb) < VALUE_BITS)
-        return AVERROR_INVALIDDATA;
-
    *h = TOP_VALUE;
    *l = 0;
    *v = get_bits_long(gb, VALUE_BITS);
-
-    return 0;
 }


--- a/libavcodec/bgmc.h
+++ b/libavcodec/bgmc.h
@@ -40,7 +40,7 @@ int ff_bgmc_init(AVCodecContext *avctx, uint8_t **cf_lut, int **cf_lut_status);
 void ff_bgmc_end(uint8_t **cf_lut, int **cf_lut_status);


-int ff_bgmc_decode_init(GetBitContext *gb,
+void ff_bgmc_decode_init(GetBitContext *gb,
                      unsigned int *h, unsigned int *l, unsigned int *v);


--- a/libavcodec/bink.c
+++ b/libavcodec/bink.c
@@ -825,7 +825,7 @@ static int binkb_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,

    binkb_init_bundles(c);
    ref_start = frame->data[plane_idx];
-    ref_end   = frame->data[plane_idx] + ((bh - 1) * frame->linesize[plane_idx] + bw - 1) * 8;
+    ref_end   = frame->data[plane_idx] + (bh * frame->linesize[plane_idx] + bw) * 8;

    for (i = 0; i < 64; i++)
        coordmap[i] = (i & 7) + (i >> 3) * stride;
@@ -879,7 +879,7 @@ static int binkb_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
                xoff = binkb_get_value(c, BINKB_SRC_X_OFF);
                yoff = binkb_get_value(c, BINKB_SRC_Y_OFF) + ybias;
                ref = dst + xoff + yoff * stride;
-                if (ref < ref_start || ref > ref_end) {
+                if (ref < ref_start || ref + 8*stride > ref_end) {
                    av_log(c->avctx, AV_LOG_WARNING, "Reference block is out of bounds\n");
                } else if (ref + 8*stride < dst || ref >= dst + 8*stride) {
                    c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
@@ -895,7 +895,7 @@ static int binkb_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
                xoff = binkb_get_value(c, BINKB_SRC_X_OFF);
                yoff = binkb_get_value(c, BINKB_SRC_Y_OFF) + ybias;
                ref = dst + xoff + yoff * stride;
-                if (ref < ref_start || ref > ref_end) {
+                if (ref < ref_start || ref + 8 * stride > ref_end) {
                    av_log(c->avctx, AV_LOG_WARNING, "Reference block is out of bounds\n");
                } else if (ref + 8*stride < dst || ref >= dst + 8*stride) {
                    c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
@@ -925,7 +925,7 @@ static int binkb_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
                xoff = binkb_get_value(c, BINKB_SRC_X_OFF);
                yoff = binkb_get_value(c, BINKB_SRC_Y_OFF) + ybias;
                ref = dst + xoff + yoff * stride;
-                if (ref < ref_start || ref > ref_end) {
+                if (ref < ref_start || ref + 8 * stride > ref_end) {
                    av_log(c->avctx, AV_LOG_WARNING, "Reference block is out of bounds\n");
                } else if (ref + 8*stride < dst || ref >= dst + 8*stride) {
                    c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
@@ -1026,7 +1026,7 @@ static int bink_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
        for (bx = 0; bx < bw; bx++, dst += 8, prev += 8) {
            blk = get_value(c, BINK_SRC_BLOCK_TYPES);
            // 16x16 block type on odd line means part of the already decoded block, so skip it
-            if (((by & 1) || (bx & 1)) && blk == SCALED_BLOCK) {
+            if ((by & 1) && blk == SCALED_BLOCK) {
                bx++;
                dst  += 8;
                prev += 8;
@@ -1299,13 +1299,13 @@ static av_cold int decode_init(AVCodecContext *avctx)
    }
    c->avctx = avctx;

-    if ((ret = av_image_check_size(avctx->width, avctx->height, 0, avctx)) < 0)
-        return ret;
-
    c->last = av_frame_alloc();
    if (!c->last)
        return AVERROR(ENOMEM);

+    if ((ret = av_image_check_size(avctx->width, avctx->height, 0, avctx)) < 0)
+        return ret;
+
    avctx->pix_fmt = c->has_alpha ? AV_PIX_FMT_YUVA420P : AV_PIX_FMT_YUV420P;

    ff_blockdsp_init(&c->bdsp, avctx);
--- a/libavcodec/binkaudio.c
+++ b/libavcodec/binkaudio.c
@@ -93,8 +93,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
    if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
        // audio is already interleaved for the RDFT format variant
        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
-        if (sample_rate > INT_MAX / avctx->channels)
-            return AVERROR_INVALIDDATA;
        sample_rate  *= avctx->channels;
        s->channels = 1;
        if (!s->version_b)
@@ -107,7 +105,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
    s->frame_len     = 1 << frame_len_bits;
    s->overlap_len   = s->frame_len / 16;
    s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
-    sample_rate_half = (sample_rate + 1LL) / 2;
+    sample_rate_half = (sample_rate + 1) / 2;
    if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
        s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
    else
--- a/libavcodec/binkdsp.c
+++ b/libavcodec/binkdsp.c
@@ -33,22 +33,20 @@
 #define A3  3784
 #define A4 -5352

-#define MUL(X,Y) ((int)((unsigned)(X) * (Y)) >> 11)
-
 #define IDCT_TRANSFORM(dest,s0,s1,s2,s3,s4,s5,s6,s7,d0,d1,d2,d3,d4,d5,d6,d7,munge,src) {\
    const int a0 = (src)[s0] + (src)[s4]; \
    const int a1 = (src)[s0] - (src)[s4]; \
    const int a2 = (src)[s2] + (src)[s6]; \
-    const int a3 = MUL(A1, (src)[s2] - (src)[s6]); \
+    const int a3 = (A1*((src)[s2] - (src)[s6])) >> 11; \
    const int a4 = (src)[s5] + (src)[s3]; \
    const int a5 = (src)[s5] - (src)[s3]; \
    const int a6 = (src)[s1] + (src)[s7]; \
    const int a7 = (src)[s1] - (src)[s7]; \
    const int b0 = a4 + a6; \
-    const int b1 = MUL(A3, a5 + a7); \
-    const int b2 = MUL(A4, a5) - b0 + b1; \
-    const int b3 = MUL(A1, a6 - a4) - b2; \
-    const int b4 = MUL(A2, a7) + b3 - b1; \
+    const int b1 = (A3*(a5 + a7)) >> 11; \
+    const int b2 = ((A4*a5) >> 11) - b0 + b1; \
+    const int b3 = (A1*(a6 - a4) >> 11) - b2; \
+    const int b4 = ((A2*a7) >> 11) + b3 - b1; \
    (dest)[d0] = munge(a0+a2   +b0); \
    (dest)[d1] = munge(a1+a3-a2+b2); \
    (dest)[d2] = munge(a1-a3+a2+b3); \
--- a/libavcodec/bintext.c
+++ b/libavcodec/bintext.c
@@ -35,8 +35,6 @@
 #include "bintext.h"
 #include "internal.h"

-#define FONT_WIDTH 8
-
 typedef struct XbinContext {
    AVFrame *frame;
    int palette[16];
@@ -63,10 +61,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
            av_log(avctx, AV_LOG_ERROR, "not enough extradata\n");
            return AVERROR_INVALIDDATA;
        }
-        if (!s->font_height) {
-            av_log(avctx, AV_LOG_ERROR, "invalid font height\n");
-            return AVERROR_INVALIDDATA;
-        }
    } else {
        s->font_height = 8;
        s->flags = 0;
@@ -97,9 +91,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
            break;
        }
    }
-    if (avctx->width < FONT_WIDTH || avctx->height < s->font_height)
-        return AVERROR_INVALIDDATA;
-

    s->frame = av_frame_alloc();
    if (!s->frame)
@@ -122,6 +113,8 @@ av_unused static void hscroll(AVCodecContext *avctx)
    }
 }

+#define FONT_WIDTH 8
+
 /**
 * Draw character to screen
 */
--- a/libavcodec/bitstream.c
+++ b/libavcodec/bitstream.c
@@ -194,9 +194,8 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
            }
            for (k = 0; k < nb; k++) {
                int bits = table[j][1];
-                int oldsym  = table[j][0];
                ff_dlog(NULL, "%4x: code=%d n=%d\n", j, i, n);
-                if ((bits || oldsym) && (bits != n || oldsym != symbol)) {
+                if (bits != 0 && bits != n) {
                    av_log(NULL, AV_LOG_ERROR, "incorrect codes\n");
                    return AVERROR_INVALIDDATA;
                }
@@ -233,10 +232,6 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
            /* note: realloc has been done, so reload tables */
            table = (volatile VLC_TYPE (*)[2])&vlc->table[table_index];
            table[j][0] = index; //code
-            if (table[j][0] != index) {
-                avpriv_request_sample(NULL, "strange codes");
-                return AVERROR_PATCHWELCOME;
-            }
            i = k-1;
        }
    }
--- a/libavcodec/bmp.c
+++ b/libavcodec/bmp.c
@@ -286,7 +286,7 @@ static int bmp_decode_frame(AVCodecContext *avctx,
        case 1:
            for (i = 0; i < avctx->height; i++) {
                int j;
-                for (j = 0; j < avctx->width >> 3; j++) {
+                for (j = 0; j < n; j++) {
                    ptr[j*8+0] =  buf[j] >> 7;
                    ptr[j*8+1] = (buf[j] >> 6) & 1;
                    ptr[j*8+2] = (buf[j] >> 5) & 1;
@@ -296,9 +296,6 @@ static int bmp_decode_frame(AVCodecContext *avctx,
                    ptr[j*8+6] = (buf[j] >> 1) & 1;
                    ptr[j*8+7] =  buf[j]       & 1;
                }
-                for (j = 0; j < (avctx->width & 7); j++) {
-                    ptr[avctx->width - (avctx->width & 7) + j] = buf[avctx->width >> 3] >> (7 - j) & 1;
-                }
                buf += n;
                ptr += linesize;
            }
--- a/libavcodec/bmp_parser.c
+++ b/libavcodec/bmp_parser.c
@@ -53,8 +53,7 @@ restart:
            if (bpc->pc.frame_start_found == 0) {
                if ((state >> 48) == (('B' << 8) | 'M')) {
                    bpc->fsize = av_bswap32(state >> 16);
-                    if (bpc->fsize > 17)
-                        bpc->pc.frame_start_found = 1;
+                    bpc->pc.frame_start_found = 1;
                }
            } else if (bpc->pc.frame_start_found == 2+4+4) {
 //                 unsigned hsize = av_bswap32(state>>32);
@@ -68,12 +67,8 @@ restart:

                if (bpc->pc.index + i > 17) {
                    next = i - 17;
-                    state = 0;
-                    break;
-                } else {
-                    bpc->pc.state64 = 0;
+                } else
                    goto restart;
-                }
            } else if (bpc->pc.frame_start_found)
                bpc->pc.frame_start_found++;
        }
@@ -94,10 +89,7 @@ flush:
    if (ff_combine_frame(&bpc->pc, next, &buf, &buf_size) < 0)
        return buf_size;

-    if (next != END_NOT_FOUND && next < 0)
-        bpc->pc.frame_start_found = FFMAX(bpc->pc.frame_start_found - i - 1, 0);
-    else
-        bpc->pc.frame_start_found = 0;
+    bpc->pc.frame_start_found = 0;

    *poutbuf      = buf;
    *poutbuf_size = buf_size;
--- a/libavcodec/bmvvideo.c
+++ b/libavcodec/bmvvideo.c
@@ -107,7 +107,7 @@ static int decode_bmv_frame(const uint8_t *source, int src_len, uint8_t *frame,
                    if (src < source || src >= source_end)
                        return AVERROR_INVALIDDATA;
                    shift += 2;
-                    val |= (unsigned)*src << shift;
+                    val |= *src << shift;
                    if (*src & 0xC)
                        break;
                }
--- a/libavcodec/brenderpix.c
+++ b/libavcodec/brenderpix.c
@@ -204,10 +204,6 @@ static int pix_decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
        avpriv_request_sample(avctx, "Format %d", hdr.format);
        return AVERROR_PATCHWELCOME;
    }
-    bytes_per_scanline = bytes_pp * hdr.width;
-
-    if (bytestream2_get_bytes_left(&gb) < hdr.height * bytes_per_scanline)
-        return AVERROR_INVALIDDATA;

    if ((ret = ff_set_dimensions(avctx, hdr.width, hdr.height)) < 0)
        return ret;
@@ -265,6 +261,7 @@ static int pix_decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
    bytestream2_skip(&gb, 8);

    // read the image data to the buffer
+    bytes_per_scanline = bytes_pp * hdr.width;
    bytes_left = bytestream2_get_bytes_left(&gb);

    if (chunk_type != IMAGE_DATA_CHUNK || data_len != bytes_left ||
--- a/libavcodec/cabac.c
+++ b/libavcodec/cabac.c
@@ -51,7 +51,7 @@ void ff_init_cabac_encoder(CABACContext *c, uint8_t *buf, int buf_size){
 *
 * @param buf_size size of buf in bits
 */
-int ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
+void ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
    c->bytestream_start=
    c->bytestream= buf;
    c->bytestream_end= buf + buf_size;
@@ -64,9 +64,6 @@ int ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
 #endif
    c->low+= ((*c->bytestream++)<<2) + 2;
    c->range= 0x1FE;
-    if ((c->range<<(CABAC_BITS+1)) < c->low)
-        return AVERROR_INVALIDDATA;
-    return 0;
 }

 void ff_init_cabac_states(void)
--- a/libavcodec/cabac.h
+++ b/libavcodec/cabac.h
@@ -56,7 +56,7 @@ typedef struct CABACContext{
 }CABACContext;

 void ff_init_cabac_encoder(CABACContext *c, uint8_t *buf, int buf_size);
-int ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size);
+void ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size);
 void ff_init_cabac_states(void);

 #endif /* AVCODEC_CABAC_H */
--- a/libavcodec/cabac_functions.h
+++ b/libavcodec/cabac_functions.h
@@ -74,8 +74,7 @@ static inline void renorm_cabac_decoder_once(CABACContext *c){

 #ifndef get_cabac_inline
 static void refill2(CABACContext *c){
-    int i;
-    unsigned x;
+    int i, x;

    x= c->low ^ (c->low-1);
    i= 7 - ff_h264_norm_shift[x>>(CABAC_BITS-1)];
@@ -191,8 +190,7 @@ static av_unused const uint8_t* skip_bytes(CABACContext *c, int n) {
 #endif
    if ((int) (c->bytestream_end - ptr) < n)
        return NULL;
-    if (ff_init_cabac_decoder(c, ptr + n, c->bytestream_end - ptr - n) < 0)
-        return NULL;
+    ff_init_cabac_decoder(c, ptr + n, c->bytestream_end - ptr - n);

    return ptr;
 }
--- a/libavcodec/cavs.c
+++ b/libavcodec/cavs.c
@@ -538,7 +538,8 @@ void ff_cavs_inter(AVSContext *h, enum cavs_mb mb_type)
 static inline void scale_mv(AVSContext *h, int *d_x, int *d_y,
                            cavs_vector *src, int distp)
 {
-    int64_t den = h->scale_den[FFMAX(src->ref, 0)];
+    int den = h->scale_den[FFMAX(src->ref, 0)];
+
    *d_x = (src->x * distp * den + 256 + FF_SIGNBIT(src->x)) >> 9;
    *d_y = (src->y * distp * den + 256 + FF_SIGNBIT(src->y)) >> 9;
 }
@@ -613,15 +614,8 @@ void ff_cavs_mv(AVSContext *h, enum cavs_mv_loc nP, enum cavs_mv_loc nC,
        mv_pred_median(h, mvP, mvA, mvB, mvC);

    if (mode < MV_PRED_PSKIP) {
-        int mx = get_se_golomb(&h->gb) + (unsigned)mvP->x;
-        int my = get_se_golomb(&h->gb) + (unsigned)mvP->y;
-
-        if (mx != (int16_t)mx || my != (int16_t)my) {
-            av_log(h->avctx, AV_LOG_ERROR, "MV %d %d out of supported range\n", mx, my);
-        } else {
-            mvP->x = mx;
-            mvP->y = my;
-        }
+        mvP->x += get_se_golomb(&h->gb);
+        mvP->y += get_se_golomb(&h->gb);
    }
    set_mvs(mvP, size);
 }
--- a/libavcodec/cavsdec.c
+++ b/libavcodec/cavsdec.c
@@ -466,7 +466,7 @@ static inline void mv_pred_direct(AVSContext *h, cavs_vector *pmv_fw,
                                  cavs_vector *col_mv)
 {
    cavs_vector *pmv_bw = pmv_fw + MV_BWD_OFFS;
-    unsigned den = h->direct_den[col_mv->ref];
+    int den = h->direct_den[col_mv->ref];
    int m = FF_SIGNBIT(col_mv->x);

    pmv_fw->dist = h->dist[1];
@@ -592,21 +592,14 @@ static int decode_residual_block(AVSContext *h, GetBitContext *gb,
 }


-static inline int decode_residual_chroma(AVSContext *h)
+static inline void decode_residual_chroma(AVSContext *h)
 {
-    if (h->cbp & (1 << 4)) {
-        int ret = decode_residual_block(h, &h->gb, chroma_dec, 0,
+    if (h->cbp & (1 << 4))
+        decode_residual_block(h, &h->gb, chroma_dec, 0,
                              ff_cavs_chroma_qp[h->qp], h->cu, h->c_stride);
-        if (ret < 0)
-            return ret;
-    }
-    if (h->cbp & (1 << 5)) {
-        int ret = decode_residual_block(h, &h->gb, chroma_dec, 0,
+    if (h->cbp & (1 << 5))
+        decode_residual_block(h, &h->gb, chroma_dec, 0,
                              ff_cavs_chroma_qp[h->qp], h->cv, h->c_stride);
-        if (ret < 0)
-            return ret;
-    }
-    return 0;
 }

 static inline int decode_residual_inter(AVSContext *h)
@@ -623,7 +616,7 @@ static inline int decode_residual_inter(AVSContext *h)

    /* get quantizer */
    if (h->cbp && !h->qp_fixed)
-        h->qp = (h->qp + (unsigned)get_se_golomb(&h->gb)) & 63;
+        h->qp = (h->qp + get_se_golomb(&h->gb)) & 63;
    for (block = 0; block < 4; block++)
        if (h->cbp & (1 << block))
            decode_residual_block(h, &h->gb, inter_dec, 0, h->qp,
@@ -657,7 +650,6 @@ static int decode_mb_i(AVSContext *h, int cbp_code)
    uint8_t top[18];
    uint8_t *left = NULL;
    uint8_t *d;
-    int ret;

    ff_cavs_init_mb(h);

@@ -701,11 +693,8 @@ static int decode_mb_i(AVSContext *h, int cbp_code)
        ff_cavs_load_intra_pred_luma(h, top, &left, block);
        h->intra_pred_l[h->pred_mode_Y[scan3x3[block]]]
            (d, top, left, h->l_stride);
-        if (h->cbp & (1<<block)) {
-            ret = decode_residual_block(h, gb, intra_dec, 1, h->qp, d, h->l_stride);
-            if (ret < 0)
-                return ret;
-        }
+        if (h->cbp & (1<<block))
+            decode_residual_block(h, gb, intra_dec, 1, h->qp, d, h->l_stride);
    }

    /* chroma intra prediction */
@@ -715,9 +704,7 @@ static int decode_mb_i(AVSContext *h, int cbp_code)
    h->intra_pred_c[pred_mode_uv](h->cv, &h->top_border_v[h->mbx * 10],
                                  h->left_border_v, h->c_stride);

-    ret = decode_residual_chroma(h);
-    if (ret < 0)
-        return ret;
+    decode_residual_chroma(h);
    ff_cavs_filter(h, I_8X8);
    set_mv_intra(h);
    return 0;
@@ -1045,10 +1032,6 @@ static int decode_pic(AVSContext *h)
    h->scale_den[1] = h->dist[1] ? 512/h->dist[1] : 0;
    if (h->cur.f->pict_type == AV_PICTURE_TYPE_B) {
        h->sym_factor = h->dist[0] * h->scale_den[1];
-        if (FFABS(h->sym_factor) > 32768) {
-            av_log(h->avctx, AV_LOG_ERROR, "sym_factor %d too large\n", h->sym_factor);
-            return AVERROR_INVALIDDATA;
-        }
    } else {
        h->direct_den[0] = h->dist[0] ? 16384 / h->dist[0] : 0;
        h->direct_den[1] = h->dist[1] ? 16384 / h->dist[1] : 0;
@@ -1081,11 +1064,6 @@ static int decode_pic(AVSContext *h)
    if (!h->loop_filter_disable && get_bits1(&h->gb)) {
        h->alpha_offset        = get_se_golomb(&h->gb);
        h->beta_offset         = get_se_golomb(&h->gb);
-        if (   h->alpha_offset < -64 || h->alpha_offset > 64
-            || h-> beta_offset < -64 || h-> beta_offset > 64) {
-            h->alpha_offset = h->beta_offset  = 0;
-            return AVERROR_INVALIDDATA;
-        }
    } else {
        h->alpha_offset = h->beta_offset  = 0;
    }
@@ -1207,7 +1185,6 @@ static int cavs_decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
    int input_size, ret;
    const uint8_t *buf_end;
    const uint8_t *buf_ptr;
-    int frame_start = 0;

    if (buf_size == 0) {
        if (!h->low_delay && h->DPB[0].f->data[0]) {
@@ -1241,11 +1218,6 @@ static int cavs_decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
                h->got_keyframe = 1;
            }
        case PIC_PB_START_CODE:
-            if (frame_start > 1)
-                return AVERROR_INVALIDDATA;
-            frame_start ++;
-            if (*got_frame)
-                av_frame_unref(data);
            *got_frame = 0;
            if (!h->got_keyframe)
                break;
--- a/libavcodec/cavsdsp.c
+++ b/libavcodec/cavsdsp.c
@@ -188,6 +188,7 @@ static void cavs_filter_ch_c(uint8_t *d, int stride, int alpha, int beta, int tc
 static void cavs_idct8_add_c(uint8_t *dst, int16_t *block, int stride) {
    int i;
    int16_t (*src)[8] = (int16_t(*)[8])block;
+    const uint8_t *cm = ff_crop_tab + MAX_NEG_CROP;

    src[0][0] += 8;

@@ -242,14 +243,14 @@ static void cavs_idct8_add_c(uint8_t *dst, int16_t *block, int stride) {
        const int b2 = a5 - a7;
        const int b3 = a4 - a6;

-        dst[i + 0*stride] = av_clip_uint8( dst[i + 0*stride] + ((b0 + b4) >> 7));
-        dst[i + 1*stride] = av_clip_uint8( dst[i + 1*stride] + ((b1 + b5) >> 7));
-        dst[i + 2*stride] = av_clip_uint8( dst[i + 2*stride] + ((b2 + b6) >> 7));
-        dst[i + 3*stride] = av_clip_uint8( dst[i + 3*stride] + ((b3 + b7) >> 7));
-        dst[i + 4*stride] = av_clip_uint8( dst[i + 4*stride] + ((b3 - b7) >> 7));
-        dst[i + 5*stride] = av_clip_uint8( dst[i + 5*stride] + ((b2 - b6) >> 7));
-        dst[i + 6*stride] = av_clip_uint8( dst[i + 6*stride] + ((b1 - b5) >> 7));
-        dst[i + 7*stride] = av_clip_uint8( dst[i + 7*stride] + ((b0 - b4) >> 7));
+        dst[i + 0*stride] = cm[ dst[i + 0*stride] + ((b0 + b4) >> 7)];
+        dst[i + 1*stride] = cm[ dst[i + 1*stride] + ((b1 + b5) >> 7)];
+        dst[i + 2*stride] = cm[ dst[i + 2*stride] + ((b2 + b6) >> 7)];
+        dst[i + 3*stride] = cm[ dst[i + 3*stride] + ((b3 + b7) >> 7)];
+        dst[i + 4*stride] = cm[ dst[i + 4*stride] + ((b3 - b7) >> 7)];
+        dst[i + 5*stride] = cm[ dst[i + 5*stride] + ((b2 - b6) >> 7)];
+        dst[i + 6*stride] = cm[ dst[i + 6*stride] + ((b1 - b5) >> 7)];
+        dst[i + 7*stride] = cm[ dst[i + 7*stride] + ((b0 - b4) >> 7)];
    }
 }

--- a/libavcodec/ccaption_dec.c
+++ b/libavcodec/ccaption_dec.c
@@ -137,9 +137,9 @@ static const uint8_t parity_table[256] = { 0, 1, 1, 0, 1, 0, 0, 1,

 struct Screen {
    /* +1 is used to compensate null character of string */
-    uint8_t characters[SCREEN_ROWS+1][SCREEN_COLUMNS+1];
-    uint8_t colors[SCREEN_ROWS+1][SCREEN_COLUMNS+1];
-    uint8_t fonts[SCREEN_ROWS+1][SCREEN_COLUMNS+1];
+    uint8_t characters[SCREEN_ROWS][SCREEN_COLUMNS+1];
+    uint8_t colors[SCREEN_ROWS][SCREEN_COLUMNS+1];
+    uint8_t fonts[SCREEN_ROWS][SCREEN_COLUMNS+1];
    /*
     * Bitmask of used rows; if a bit is not set, the
     * corresponding row is not used.
@@ -168,8 +168,7 @@ typedef struct CCaptionSubContext {
    int64_t end_time;
    char prev_cmd[2];
    /* buffer to store pkt data */
-    uint8_t *pktbuf;
-    int pktbuf_size;
+    AVBufferRef *pktbuf;
 }CCaptionSubContext;


@@ -186,7 +185,11 @@ static av_cold int init_decoder(AVCodecContext *avctx)
    if(ret < 0) {
        return ret;
    }
-
+    /* allocate pkt buffer */
+    ctx->pktbuf = av_buffer_alloc(128);
+    if( !ctx->pktbuf) {
+        ret = AVERROR(ENOMEM);
+    }
    return ret;
 }

@@ -194,8 +197,7 @@ static av_cold int close_decoder(AVCodecContext *avctx)
 {
    CCaptionSubContext *ctx = avctx->priv_data;
    av_bprint_finalize( &ctx->buffer, NULL);
-    av_freep(&ctx->pktbuf);
-    ctx->pktbuf_size = 0;
+    av_buffer_unref(&ctx->pktbuf);
    return 0;
 }

@@ -527,13 +529,16 @@ static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avp
    int ret = 0;
    int i;

-    av_fast_padded_malloc(&ctx->pktbuf, &ctx->pktbuf_size, len);
-    if (!ctx->pktbuf) {
-        av_log(ctx, AV_LOG_WARNING, "Insufficient Memory of %d truncated to %d\n", len, ctx->pktbuf_size);
-        return AVERROR(ENOMEM);
+    if ( ctx->pktbuf->size < len) {
+        ret = av_buffer_realloc(&ctx->pktbuf, len);
+         if(ret < 0) {
+            av_log(ctx, AV_LOG_WARNING, "Insufficient Memory of %d truncated to %d\n",len, ctx->pktbuf->size);
+            len = ctx->pktbuf->size;
+            ret = 0;
+        }
    }
-    memcpy(ctx->pktbuf, avpkt->data, len);
-    bptr = ctx->pktbuf;
+    memcpy(ctx->pktbuf->data, avpkt->data, len);
+    bptr = ctx->pktbuf->data;


    for (i  = 0; i < len; i += 3) {
--- a/libavcodec/cdgraphics.c
+++ b/libavcodec/cdgraphics.c
@@ -77,8 +77,11 @@ static av_cold int cdg_decode_init(AVCodecContext *avctx)
    if (!cc->frame)
        return AVERROR(ENOMEM);

+    avctx->width   = CDG_FULL_WIDTH;
+    avctx->height  = CDG_FULL_HEIGHT;
    avctx->pix_fmt = AV_PIX_FMT_PAL8;
-    return ff_set_dimensions(avctx, CDG_FULL_WIDTH, CDG_FULL_HEIGHT);
+
+    return 0;
 }

 static void cdg_border_preset(CDGraphicsContext *cc, uint8_t *data)
@@ -233,7 +236,7 @@ static void cdg_scroll(CDGraphicsContext *cc, uint8_t *data,
    for (y = FFMAX(0, vinc); y < FFMIN(CDG_FULL_HEIGHT + vinc, CDG_FULL_HEIGHT); y++)
        memcpy(out + FFMAX(0, hinc) + stride * y,
               in + FFMAX(0, hinc) - hinc + (y - vinc) * stride,
-               FFABS(stride) - FFABS(hinc));
+               FFMIN(stride + hinc, stride));

    if (vinc > 0)
        cdg_fill_wrapper(0, 0, out,
--- a/libavcodec/cdxl.c
+++ b/libavcodec/cdxl.c
@@ -252,11 +252,11 @@ static int cdxl_decode_frame(AVCodecContext *avctx, void *data,

    aligned_width = FFALIGN(c->avctx->width, 16);
    c->padded_bits  = aligned_width - c->avctx->width;
-    if (c->video_size < aligned_width * avctx->height * (int64_t)c->bpp / 8)
+    if (c->video_size < aligned_width * avctx->height * c->bpp / 8)
        return AVERROR_INVALIDDATA;
-    if (!encoding && c->palette_size && c->bpp <= 8 && c->format != CHUNKY) {
+    if (!encoding && c->palette_size && c->bpp <= 8) {
        avctx->pix_fmt = AV_PIX_FMT_PAL8;
-    } else if (encoding == 1 && (c->bpp == 6 || c->bpp == 8) && c->format != CHUNKY) {
+    } else if (encoding == 1 && (c->bpp == 6 || c->bpp == 8)) {
        if (c->palette_size != (1 << (c->bpp - 1)))
            return AVERROR_INVALIDDATA;
        avctx->pix_fmt = AV_PIX_FMT_BGR24;
--- a/libavcodec/celp_math.h
+++ b/libavcodec/celp_math.h
@@ -68,7 +68,7 @@ int ff_log2_q15(uint32_t value);
 *
 * @return value << offset, if offset>=0; value >> -offset - otherwise
 */
-static inline unsigned bidir_sal(unsigned value, int offset)
+static inline int bidir_sal(int value, int offset)
 {
    if(offset < 0) return value >> -offset;
    else           return value <<  offset;
--- a/libavcodec/cinepak.c
+++ b/libavcodec/cinepak.c
@@ -315,11 +315,17 @@ static int cinepak_decode_strip (CinepakContext *s,
    return AVERROR_INVALIDDATA;
 }

-static int cinepak_predecode_check (CinepakContext *s)
+static int cinepak_decode (CinepakContext *s)
 {
-    int           num_strips;
+    const uint8_t  *eod = (s->data + s->size);
+    int           i, result, strip_size, frame_flags, num_strips;
+    int           y0 = 0;
    int           encoded_buf_size;

+    if (s->size < 10)
+        return AVERROR_INVALIDDATA;
+
+    frame_flags = s->data[0];
    num_strips  = AV_RB16 (&s->data[8]);
    encoded_buf_size = AV_RB24(&s->data[1]);

@@ -350,21 +356,6 @@ static int cinepak_predecode_check (CinepakContext *s)
            s->sega_film_skip_bytes = 0;
    }

-    if (s->size < 10 + s->sega_film_skip_bytes + num_strips * 12)
-        return AVERROR_INVALIDDATA;
-
-    return 0;
-}
-
-static int cinepak_decode (CinepakContext *s)
-{
-    const uint8_t  *eod = (s->data + s->size);
-    int           i, result, strip_size, frame_flags, num_strips;
-    int           y0 = 0;
-
-    frame_flags = s->data[0];
-    num_strips  = AV_RB16 (&s->data[8]);
-
    s->data += 10 + s->sega_film_skip_bytes;

    num_strips = FFMIN(num_strips, MAX_STRIPS);
@@ -444,36 +435,18 @@ static int cinepak_decode_frame(AVCodecContext *avctx,
    const uint8_t *buf = avpkt->data;
    int ret = 0, buf_size = avpkt->size;
    CinepakContext *s = avctx->priv_data;
-    int num_strips;

    s->data = buf;
    s->size = buf_size;

-    if (s->size < 10)
-        return AVERROR_INVALIDDATA;
-
-    num_strips = AV_RB16 (&s->data[8]);
-
-    //Empty frame, do not waste time
-    if (!num_strips && (!s->palette_video || !av_packet_get_side_data(avpkt, AV_PKT_DATA_PALETTE, NULL)))
-        return buf_size;
-
-    if ((ret = cinepak_predecode_check(s)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "cinepak_predecode_check failed\n");
-        return ret;
-    }
-
    if ((ret = ff_reget_buffer(avctx, s->frame)) < 0)
        return ret;

    if (s->palette_video) {
-        int size;
-        const uint8_t *pal = av_packet_get_side_data(avpkt, AV_PKT_DATA_PALETTE, &size);
-        if (pal && size == AVPALETTE_SIZE) {
+        const uint8_t *pal = av_packet_get_side_data(avpkt, AV_PKT_DATA_PALETTE, NULL);
+        if (pal) {
            s->frame->palette_has_changed = 1;
            memcpy(s->pal, pal, AVPALETTE_SIZE);
-        } else if (pal) {
-            av_log(avctx, AV_LOG_ERROR, "Palette size %d is wrong\n", size);
        }
    }

--- a/libavcodec/cllc.c
+++ b/libavcodec/cllc.c
@@ -29,10 +29,6 @@
 #include "avcodec.h"
 #include "internal.h"

-#define VLC_BITS 7
-#define VLC_DEPTH 2
-
-
 typedef struct CLLCContext {
    AVCodecContext *avctx;
    BswapDSPContext bdsp;
@@ -55,13 +51,6 @@ static int read_code_table(CLLCContext *ctx, GetBitContext *gb, VLC *vlc)

    num_lens = get_bits(gb, 5);

-    if (num_lens > VLC_BITS * VLC_DEPTH) {
-        vlc->table = NULL;
-
-        av_log(ctx->avctx, AV_LOG_ERROR, "To long VLCs %d\n", num_lens);
-        return AVERROR_INVALIDDATA;
-    }
-
    for (i = 0; i < num_lens; i++) {
        num_codes      = get_bits(gb, 9);
        num_codes_sum += num_codes;
@@ -81,15 +70,11 @@ static int read_code_table(CLLCContext *ctx, GetBitContext *gb, VLC *vlc)

            count++;
        }
-        if (prefix > (65535 - 256)/2) {
-            vlc->table = NULL;
-            return AVERROR_INVALIDDATA;
-        }

        prefix <<= 1;
    }

-    return ff_init_vlc_sparse(vlc, VLC_BITS, count, bits, 1, 1,
+    return ff_init_vlc_sparse(vlc, 7, count, bits, 1, 1,
                              codes, 2, 2, symbols, 1, 1, 0);
 }

@@ -116,7 +101,7 @@ static int read_argb_line(CLLCContext *ctx, GetBitContext *gb, int *top_left,
    for (i = 0; i < ctx->avctx->width; i++) {
        /* Always get the alpha component */
        UPDATE_CACHE(bits, gb);
-        GET_VLC(code, bits, gb, vlc[0].table, VLC_BITS, VLC_DEPTH);
+        GET_VLC(code, bits, gb, vlc[0].table, 7, 2);

        pred[0] += code;
        dst[0]   = pred[0];
@@ -125,21 +110,21 @@ static int read_argb_line(CLLCContext *ctx, GetBitContext *gb, int *top_left,
        if (dst[0]) {
            /* Red */
            UPDATE_CACHE(bits, gb);
-            GET_VLC(code, bits, gb, vlc[1].table, VLC_BITS, VLC_DEPTH);
+            GET_VLC(code, bits, gb, vlc[1].table, 7, 2);

            pred[1] += code;
            dst[1]   = pred[1];

            /* Green */
            UPDATE_CACHE(bits, gb);
-            GET_VLC(code, bits, gb, vlc[2].table, VLC_BITS, VLC_DEPTH);
+            GET_VLC(code, bits, gb, vlc[2].table, 7, 2);

            pred[2] += code;
            dst[2]   = pred[2];

            /* Blue */
            UPDATE_CACHE(bits, gb);
-            GET_VLC(code, bits, gb, vlc[3].table, VLC_BITS, VLC_DEPTH);
+            GET_VLC(code, bits, gb, vlc[3].table, 7, 2);

            pred[3] += code;
            dst[3]   = pred[3];
@@ -181,7 +166,7 @@ static int read_rgb24_component_line(CLLCContext *ctx, GetBitContext *gb,
    /* Simultaneously read and restore the line */
    for (i = 0; i < ctx->avctx->width; i++) {
        UPDATE_CACHE(bits, gb);
-        GET_VLC(code, bits, gb, vlc->table, VLC_BITS, VLC_DEPTH);
+        GET_VLC(code, bits, gb, vlc->table, 7, 2);

        pred  += code;
        dst[0] = pred;
@@ -210,7 +195,7 @@ static int read_yuv_component_line(CLLCContext *ctx, GetBitContext *gb,
    /* Simultaneously read and restore the line */
    for (i = 0; i < ctx->avctx->width >> is_chroma; i++) {
        UPDATE_CACHE(bits, gb);
-        GET_VLC(code, bits, gb, vlc->table, VLC_BITS, VLC_DEPTH);
+        GET_VLC(code, bits, gb, vlc->table, 7, 2);

        pred     += code;
        outbuf[i] = pred;
--- a/libavcodec/cngdec.c
+++ b/libavcodec/cngdec.c
@@ -146,7 +146,7 @@ static int cng_decode_frame(AVCodecContext *avctx, void *data,
        return ret;
    buf_out = (int16_t *)frame->data[0];
    for (i = 0; i < avctx->frame_size; i++)
-        buf_out[i] = av_clip_int16(p->filter_out[i + p->order]);
+        buf_out[i] = p->filter_out[i + p->order];
    memcpy(p->filter_out, p->filter_out + avctx->frame_size,
           p->order * sizeof(*p->filter_out));

@@ -167,5 +167,5 @@ AVCodec ff_comfortnoise_decoder = {
    .close          = cng_decode_close,
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                     AV_SAMPLE_FMT_NONE },
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1,
 };
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -143,7 +143,7 @@ typedef struct cook {

    /* generatable tables and related variables */
    int                 gain_size_factor;
-    float               gain_table[31];
+    float               gain_table[23];

    /* data buffers */

@@ -178,8 +178,8 @@ static av_cold void init_gain_table(COOKContext *q)
 {
    int i;
    q->gain_size_factor = q->samples_per_channel / 8;
-    for (i = 0; i < 31; i++)
-        q->gain_table[i] = pow(pow2tab[i + 48],
+    for (i = 0; i < 23; i++)
+        q->gain_table[i] = pow(pow2tab[i + 52],
                               (1.0 / (double) q->gain_size_factor));
 }

@@ -663,7 +663,7 @@ static void interpolate_float(COOKContext *q, float *buffer,
        for (i = 0; i < q->gain_size_factor; i++)
            buffer[i] *= fc1;
    } else {                                        // smooth gain
-        fc2 = q->gain_table[15 + (gain_index_next - gain_index)];
+        fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
        for (i = 0; i < q->gain_size_factor; i++) {
            buffer[i] *= fc1;
            fc1       *= fc2;
@@ -752,7 +752,7 @@ static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
        for (i = 0; i < length; i++)
            decouple_tab[start + i] = get_vlc2(&q->gb,
                                               p->channel_coupling.table,
-                                               p->channel_coupling.bits, 3);
+                                               p->channel_coupling.bits, 2);
    else
        for (i = 0; i < length; i++) {
            int v = get_bits(&q->gb, p->js_vlc_bits);
@@ -1068,19 +1068,12 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
        return AVERROR_INVALIDDATA;
    }

-    if (avctx->block_align >= INT_MAX / 8)
-        return AVERROR(EINVAL);
-
    /* Initialize RNG. */
    av_lfg_init(&q->random_state, 0);

    ff_audiodsp_init(&q->adsp);

    while (edata_ptr < edata_ptr_end) {
-        if (s >= FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
-            avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
-            return AVERROR_PATCHWELCOME;
-        }
        /* 8 for mono, 16 for stereo, ? for multichannel
           Swap to right endianness so we don't need to care later on. */
        if (extradata_size >= 8) {
@@ -1220,6 +1213,10 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)

        q->num_subpackets++;
        s++;
+        if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
+            avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
+            return AVERROR_PATCHWELCOME;
+        }
    }
    /* Generate tables */
    init_pow2table();
@@ -1229,6 +1226,10 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
    if ((ret = init_cook_vlc_tables(q)))
        return ret;

+
+    if (avctx->block_align >= UINT_MAX / 2)
+        return AVERROR(EINVAL);
+
    /* Pad the databuffer with:
       DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
       AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
--- a/libavcodec/cpia.c
+++ b/libavcodec/cpia.c
@@ -63,7 +63,7 @@ static int cpia_decode_frame(AVCodecContext *avctx,
    uint8_t *y, *u, *v, *y_end, *u_end, *v_end;

    // Check header
-    if ( avpkt->size < FRAME_HEADER_SIZE + avctx->height * 3
+    if ( avpkt->size < FRAME_HEADER_SIZE
      || header[0] != MAGIC_0 || header[1] != MAGIC_1
      || (header[17] != SUBSAMPLE_420 && header[17] != SUBSAMPLE_422)
      || (header[18] != YUVORDER_YUYV && header[18] != YUVORDER_UYVY)
@@ -111,7 +111,6 @@ static int cpia_decode_frame(AVCodecContext *avctx,
        // Read line length, two byte little endian
        linelength = AV_RL16(src);
        src += 2;
-        src_size -= 2;

        if (src_size < linelength) {
            av_frame_set_decode_error_flags(frame, FF_DECODE_ERROR_INVALID_BITSTREAM);
--- a/libavcodec/cscd.c
+++ b/libavcodec/cscd.c
@@ -81,19 +81,15 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
    switch ((buf[0] >> 1) & 7) {
        case 0: { // lzo compression
            int outlen = c->decomp_size, inlen = buf_size - 2;
-            if (av_lzo1x_decode(c->decomp_buf, &outlen, &buf[2], &inlen) || outlen) {
+            if (av_lzo1x_decode(c->decomp_buf, &outlen, &buf[2], &inlen))
                av_log(avctx, AV_LOG_ERROR, "error during lzo decompression\n");
-                return AVERROR_INVALIDDATA;
-            }
            break;
        }
        case 1: { // zlib compression
 #if CONFIG_ZLIB
            unsigned long dlen = c->decomp_size;
-            if (uncompress(c->decomp_buf, &dlen, &buf[2], buf_size - 2) != Z_OK) {
+            if (uncompress(c->decomp_buf, &dlen, &buf[2], buf_size - 2) != Z_OK)
                av_log(avctx, AV_LOG_ERROR, "error during zlib decompression\n");
-                return AVERROR_INVALIDDATA;
-            }
            break;
 #else
            av_log(avctx, AV_LOG_ERROR, "compiled without zlib support\n");
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -41,6 +41,8 @@ int avpriv_dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst,
 {
    uint32_t mrk;
    int i, tmp;
+    const uint16_t *ssrc = (const uint16_t *) src;
+    uint16_t *sdst = (uint16_t *) dst;
    PutBitContext pb;

    if ((unsigned) src_size > (unsigned) max_size)
@@ -52,11 +54,8 @@ int avpriv_dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst,
        memcpy(dst, src, src_size);
        return src_size;
    case DCA_SYNCWORD_CORE_LE:
-        for (i = 0; i < (src_size + 1) >> 1; i++) {
-            AV_WB16(dst, AV_RL16(src));
-            src += 2;
-            dst += 2;
-        }
+        for (i = 0; i < (src_size + 1) >> 1; i++)
+            *sdst++ = av_bswap16(*ssrc++);
        return src_size;
    case DCA_SYNCWORD_CORE_14B_BE:
    case DCA_SYNCWORD_CORE_14B_LE:
--- a/libavcodec/dds.c
+++ b/libavcodec/dds.c
@@ -38,7 +38,7 @@

 #define DDPF_FOURCC    (1 <<  2)
 #define DDPF_PALETTE   (1 <<  5)
-#define DDPF_NORMALMAP (1U << 31)
+#define DDPF_NORMALMAP (1 << 31)

 enum DDSPostProc {
    DDS_NONE = 0,
@@ -141,12 +141,6 @@ static int parse_pixel_format(AVCodecContext *avctx)
    normal_map      = flags & DDPF_NORMALMAP;
    fourcc = bytestream2_get_le32(gbc);

-    if (ctx->compressed && ctx->paletted) {
-        av_log(avctx, AV_LOG_WARNING,
-               "Disabling invalid palette flag for compressed dds.\n");
-        ctx->paletted = 0;
-    }
-
    bpp = bytestream2_get_le32(gbc); // rgbbitcount
    r   = bytestream2_get_le32(gbc); // rbitmask
    g   = bytestream2_get_le32(gbc); // gbitmask
@@ -600,7 +594,6 @@ static int dds_decode(AVCodecContext *avctx, void *data,
    AVFrame *frame = data;
    int mipmap;
    int ret;
-    int width, height;

    ff_texturedsp_init(&ctx->texdsp);
    bytestream2_init(gbc, avpkt->data, avpkt->size);
@@ -619,9 +612,9 @@ static int dds_decode(AVCodecContext *avctx, void *data,

    bytestream2_skip(gbc, 4); // flags

-    height = bytestream2_get_le32(gbc);
-    width  = bytestream2_get_le32(gbc);
-    ret = ff_set_dimensions(avctx, width, height);
+    avctx->height = bytestream2_get_le32(gbc);
+    avctx->width  = bytestream2_get_le32(gbc);
+    ret = av_image_check_size(avctx->width, avctx->height, 0, avctx);
    if (ret < 0) {
        av_log(avctx, AV_LOG_ERROR, "Invalid image size %dx%d.\n",
               avctx->width, avctx->height);
@@ -649,18 +642,9 @@ static int dds_decode(AVCodecContext *avctx, void *data,
        return ret;

    if (ctx->compressed) {
-        int size = (avctx->coded_height / TEXTURE_BLOCK_H) *
-                   (avctx->coded_width / TEXTURE_BLOCK_W) * ctx->tex_ratio;
        ctx->slice_count = av_clip(avctx->thread_count, 1,
                                   avctx->coded_height / TEXTURE_BLOCK_H);

-        if (bytestream2_get_bytes_left(gbc) < size) {
-            av_log(avctx, AV_LOG_ERROR,
-                   "Compressed Buffer is too small (%d < %d).\n",
-                   bytestream2_get_bytes_left(gbc), size);
-            return AVERROR_INVALIDDATA;
-        }
-
        /* Use the decompress function on the texture, one block per thread. */
        ctx->tex_data = gbc->buffer;
        avctx->execute2(avctx, decompress_texture_thread, frame, NULL, ctx->slice_count);
@@ -676,18 +660,12 @@ static int dds_decode(AVCodecContext *avctx, void *data,
                        (frame->data[1][2+i*4]<<0)+
                        (frame->data[1][1+i*4]<<8)+
                        (frame->data[1][0+i*4]<<16)+
-                        ((unsigned)frame->data[1][3+i*4]<<24)
+                        (frame->data[1][3+i*4]<<24)
                );

            frame->palette_has_changed = 1;
        }

-        if (bytestream2_get_bytes_left(gbc) < frame->height * linesize) {
-            av_log(avctx, AV_LOG_ERROR, "Buffer is too small (%d < %d).\n",
-                   bytestream2_get_bytes_left(gbc), frame->height * linesize);
-            return AVERROR_INVALIDDATA;
-        }
-
        av_image_copy_plane(frame->data[0], frame->linesize[0],
                            gbc->buffer, linesize,
                            linesize, frame->height);
--- a/Show More
+++ b/Show More
Author	SHA1	Message	Date
Hendrik Leppkes	cebe51a625	Merge commit '9d58639e270f7612874681e0ca30fa461e2667b7' * commit '9d58639e270f7612874681e0ca30fa461e2667b7': sws: Drop deprecated SWS_CPU_CAPS defines Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>	2015-09-05 15:58:52 +02:00
Vittorio Giovara	9d58639e27	sws: Drop deprecated SWS_CPU_CAPS defines Deprecated in 07/2012.	2015-08-28 10:55:26 +02:00
@@ -1 +1 @@
 .8.22
 .7.git