@scrypted/libav 1.0.170 → 1.0.171

package/opus/doc/opus_logo.svg

@@ -1,157 +0,0 @@
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package/opus/doc/opus_update.patch

@@ -1,244 +0,0 @@
-diff --git a/celt/bands.c b/celt/bands.c
-index 6962587..32e1de6 100644
---- a/celt/bands.c
-+++ b/celt/bands.c
-@@ -1234,9 +1234,23 @@ void quant_all_bands(int encode, const CELTMode *m, int start, int end,
-          b = 0;
-       }
- 
--      if (resynth && M*eBands[i]-N >= M*eBands[start] && (update_lowband || lowband_offset==0))
-+      if (resynth && (M*eBands[i]-N >= M*eBands[start] || i==start+1) && (update_lowband || lowband_offset==0))
-             lowband_offset = i;
- 
-+      if (i == start+1)
-+      {
-+         int n1, n2;
-+         int offset;
-+         n1 = M*(eBands[start+1]-eBands[start]);
-+         n2 = M*(eBands[start+2]-eBands[start+1]);
-+         offset = M*eBands[start];
-+         /* Duplicate enough of the first band folding data to be able to fold the second band.
-+            Copies no data for CELT-only mode. */
-+         OPUS_COPY(&norm[offset+n1], &norm[offset+2*n1 - n2], n2-n1);
-+         if (C==2)
-+            OPUS_COPY(&norm2[offset+n1], &norm2[offset+2*n1 - n2], n2-n1);
-+      }
-+
-       tf_change = tf_res[i];
-       if (i>=m->effEBands)
-       {
-@@ -1257,7 +1271,7 @@ void quant_all_bands(int encode, const CELTMode *m, int start, int end,
-          fold_start = lowband_offset;
-          while(M*eBands[--fold_start] > effective_lowband);
-          fold_end = lowband_offset-1;
--         while(M*eBands[++fold_end] < effective_lowband+N);
-+         while(++fold_end < i && M*eBands[fold_end] < effective_lowband+N);
-          x_cm = y_cm = 0;
-          fold_i = fold_start; do {
-            x_cm |= collapse_masks[fold_i*C+0];
-diff --git a/celt/quant_bands.c b/celt/quant_bands.c
-index e5ed9ef..82fb823 100644
---- a/celt/quant_bands.c
-+++ b/celt/quant_bands.c
-@@ -552,6 +552,7 @@ void log2Amp(const CELTMode *m, int start, int end,
-       {
-          opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands],
-                          SHL16((opus_val16)eMeans[i],6));
-+         lg = MIN32(QCONST32(32.f, 16), lg);
-          eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4);
-       }
-       for (;i<m->nbEBands;i++)
-diff --git a/silk/LPC_inv_pred_gain.c b/silk/LPC_inv_pred_gain.c
-index 60c439b..6c301da 100644
---- a/silk/LPC_inv_pred_gain.c
-+++ b/silk/LPC_inv_pred_gain.c
-@@ -84,8 +84,13 @@ static opus_int32 LPC_inverse_pred_gain_QA(                 /* O   Returns inver
- 
-         /* Update AR coefficient */
-         for( n = 0; n < k; n++ ) {
--            tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( Aold_QA[ k - n - 1 ], rc_Q31, 31 );
--            Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q );
-+            opus_int64 tmp64;
-+            tmp_QA = silk_SUB_SAT32( Aold_QA[ n ], MUL32_FRAC_Q( Aold_QA[ k - n - 1 ], rc_Q31, 31 ) );
-+            tmp64 = silk_RSHIFT_ROUND64( silk_SMULL( tmp_QA, rc_mult2 ), mult2Q);
-+            if( tmp64 > silk_int32_MAX || tmp64 < silk_int32_MIN ) {
-+               return 0;
-+            }
-+            Anew_QA[ n ] = ( opus_int32 )tmp64;
-         }
-     }
- 
-diff --git a/silk/NLSF_stabilize.c b/silk/NLSF_stabilize.c
-index 979aaba..2ef2398 100644
---- a/silk/NLSF_stabilize.c
-+++ b/silk/NLSF_stabilize.c
-@@ -134,7 +134,7 @@ void silk_NLSF_stabilize(
- 
-         /* Keep delta_min distance between the NLSFs */
-         for( i = 1; i < L; i++ )
--            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], NLSF_Q15[i-1] + NDeltaMin_Q15[i] );
-+            NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], silk_ADD_SAT16( NLSF_Q15[i-1], NDeltaMin_Q15[i] ) );
- 
-         /* Last NLSF should be no higher than 1 - NDeltaMin[L] */
-         NLSF_Q15[L-1] = silk_min_int( NLSF_Q15[L-1], (1<<15) - NDeltaMin_Q15[L] );
-diff --git a/silk/dec_API.c b/silk/dec_API.c
-index efd7918..21bb7e0 100644
---- a/silk/dec_API.c
-+++ b/silk/dec_API.c
-@@ -72,6 +72,9 @@ opus_int silk_InitDecoder(                              /* O    Returns error co
-     for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
-         ret  = silk_init_decoder( &channel_state[ n ] );
-     }
-+    silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo));
-+    /* Not strictly needed, but it's cleaner that way */
-+    ((silk_decoder *)decState)->prev_decode_only_middle = 0;
- 
-     return ret;
- }
-diff --git a/silk/resampler_private_IIR_FIR.c b/silk/resampler_private_IIR_FIR.c
-index dbd6d9a..91a43aa 100644
---- a/silk/resampler_private_IIR_FIR.c
-+++ b/silk/resampler_private_IIR_FIR.c
-@@ -75,10 +75,10 @@ void silk_resampler_private_IIR_FIR(
-     silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
-     opus_int32 nSamplesIn;
-     opus_int32 max_index_Q16, index_increment_Q16;
--    opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ];
-+    opus_int16 buf[ 2*RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ];
- 
-     /* Copy buffered samples to start of buffer */
--    silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
-+    silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
- 
-     /* Iterate over blocks of frameSizeIn input samples */
-     index_increment_Q16 = S->invRatio_Q16;
-@@ -95,13 +95,13 @@ void silk_resampler_private_IIR_FIR(
- 
-         if( inLen > 0 ) {
-             /* More iterations to do; copy last part of filtered signal to beginning of buffer */
--            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
-+            silk_memmove( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
-         } else {
-             break;
-         }
-     }
- 
-     /* Copy last part of filtered signal to the state for the next call */
--    silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
-+    silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
- }
- 
-diff --git a/src/opus_decoder.c b/src/opus_decoder.c
-index 0cc56f8..8a30fbc 100644
---- a/src/opus_decoder.c
-+++ b/src/opus_decoder.c
-@@ -595,16 +595,14 @@ static int opus_packet_parse_impl(const unsigned char *data, int len,
-       /* Padding flag is bit 6 */
-       if (ch&0x40)
-       {
--         int padding=0;
-          int p;
-          do {
-             if (len<=0)
-                return OPUS_INVALID_PACKET;
-             p = *data++;
-             len--;
--            padding += p==255 ? 254: p;
-+            len -= p==255 ? 254: p;
-          } while (p==255);
--         len -= padding;
-       }
-       if (len<0)
-          return OPUS_INVALID_PACKET;
-diff --git a/run_vectors.sh b/run_vectors.sh
-index 7cd23ed..4841b0a 100755
---- a/run_vectors.sh
-+++ b/run_vectors.sh
-@@ -1,3 +1,5 @@
-+#!/bin/sh
-+#
- # Copyright (c) 2011-2012 IETF Trust, Jean-Marc Valin. All rights reserved.
- #
- #  This file is extracted from RFC6716. Please see that RFC for additional
-@@ -31,10 +33,8 @@
- #  NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- #  SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- 
--#!/bin/sh
--
--rm logs_mono.txt
--rm logs_stereo.txt
-+rm -f logs_mono.txt logs_mono2.txt
-+rm -f logs_stereo.txt logs_stereo2.txt
- 
- if [ "$#" -ne "3" ]; then
-     echo "usage: run_vectors.sh <exec path> <vector path> <rate>"
-@@ -45,18 +45,23 @@ CMD_PATH=$1
- VECTOR_PATH=$2
- RATE=$3
- 
--OPUS_DEMO=$CMD_PATH/opus_demo
--OPUS_COMPARE=$CMD_PATH/opus_compare
-+: ${OPUS_DEMO:=$CMD_PATH/opus_demo}
-+: ${OPUS_COMPARE:=$CMD_PATH/opus_compare}
- 
- if [ -d $VECTOR_PATH ]; then
-     echo Test vectors found in $VECTOR_PATH
- else
-     echo No test vectors found
--    #Don't make the test fail here because the test vectors will be
--    #distributed separately
-+    #Don't make the test fail here because the test vectors
-+    #will be distributed separately
-     exit 0
- fi
- 
-+if [ ! -x $OPUS_COMPARE ]; then
-+    echo ERROR: Compare program not found: $OPUS_COMPARE
-+    exit 1
-+fi
-+
- if [ -x $OPUS_DEMO ]; then
-     echo Decoding with $OPUS_DEMO
- else
-@@ -82,9 +87,11 @@ do
-         echo ERROR: decoding failed
-         exit 1
-     fi
--    $OPUS_COMPARE -r $RATE $VECTOR_PATH/testvector$file.dec tmp.out >> logs_mono.txt 2>&1
-+    $OPUS_COMPARE -r $RATE $VECTOR_PATH/testvector${file}.dec tmp.out >> logs_mono.txt 2>&1
-     float_ret=$?
--    if [ "$float_ret" -eq "0" ]; then
-+    $OPUS_COMPARE -r $RATE $VECTOR_PATH/testvector${file}m.dec tmp.out >> logs_mono2.txt 2>&1
-+    float_ret2=$?
-+    if [ "$float_ret" -eq "0" ] || [ "$float_ret2" -eq "0" ]; then
-         echo output matches reference
-     else
-         echo ERROR: output does not match reference
-@@ -111,9 +118,11 @@ do
-         echo ERROR: decoding failed
-         exit 1
-     fi
--    $OPUS_COMPARE -s -r $RATE $VECTOR_PATH/testvector$file.dec tmp.out >> logs_stereo.txt 2>&1
-+    $OPUS_COMPARE -s -r $RATE $VECTOR_PATH/testvector${file}.dec tmp.out >> logs_stereo.txt 2>&1
-     float_ret=$?
--    if [ "$float_ret" -eq "0" ]; then
-+    $OPUS_COMPARE -s -r $RATE $VECTOR_PATH/testvector${file}m.dec tmp.out >> logs_stereo2.txt 2>&1
-+    float_ret2=$?
-+    if [ "$float_ret" -eq "0" ] || [ "$float_ret2" -eq "0" ]; then
-         echo output matches reference
-     else
-         echo ERROR: output does not match reference
-@@ -125,5 +134,10 @@ done
- 
- 
- echo All tests have passed successfully
--grep quality logs_mono.txt | awk '{sum+=$4}END{print "Average mono quality is", sum/NR, "%"}'
--grep quality logs_stereo.txt | awk '{sum+=$4}END{print "Average stereo quality is", sum/NR, "%"}'
-+mono1=`grep quality logs_mono.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'`
-+mono2=`grep quality logs_mono2.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'`
-+echo $mono1 $mono2 | awk '{if ($2 > $1) $1 = $2; print "Average mono quality is", $1, "%"}'
-+
-+stereo1=`grep quality logs_stereo.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'`
-+stereo2=`grep quality logs_stereo2.txt | awk '{sum+=$4}END{if (NR == 12) sum /= 12; else sum = 0; print sum}'`
-+echo $stereo1 $stereo2 | awk '{if ($2 > $1) $1 = $2; print "Average stereo quality is", $1, "%"}'

package/opus/doc/release.txt

@@ -1,43 +0,0 @@
-= Release checklist =
-
-== Source release ==
-
-- Check for uncommitted changes to master.
-- Update OPUS_LT_* API versioning in configure.ac.
-- Tag the release commit with 'git tag -s vN.M'.
- - Include release notes in the tag annotation.
-- Verify 'make distcheck' produces a tarball with
-  the desired name.
-- Push tag to public repo.
-- Upload source package 'opus-${version}.tar.gz'
- - Add to https://svn.xiph.org/releases/opus/
- - Update checksum files
- - svn commit
- - Copy to archive.mozilla.org/pub/opus/
- - Update checksum files there as well.
-- Add release notes to https://gitlab.xiph.org/xiph/opus-website.git
-- Update links and checksums on the downloads page.
-- Add a copy of the documentation to <https://www.opus-codec.org/docs/>
-  and update the links.
-- Update /topic in #opus IRC channel.
-
-Releases are committed to https://svn.xiph.org/releases/opus/
-which propagates to downloads.xiph.org, and copied manually
-to https://archive.mozilla.org/pub/opus/
-
-Website updates are committed to https://gitlab.xiph.org/xiph/opus-website.git
-which propagates to https://opus-codec.org/
-
-== Binary release ==
-
-We usually build opus-tools binaries for MacOS and Windows.
-
-Binary releases are copied manually to
-https://archive.mozilla.org/pub/opus/win32/
-
-For Mac, submit a pull request to homebrew.
-
-== Website updates ==
-
-For major releases, recreate the files on https://opus-codec.org/examples/
-with the next encoder.

package/opus/doc/trivial_example.c

@@ -1,171 +0,0 @@
-/* Copyright (c) 2013 Jean-Marc Valin */
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
-   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-/* This is meant to be a simple example of encoding and decoding audio
-   using Opus. It should make it easy to understand how the Opus API
-   works. For more information, see the full API documentation at:
-   https://www.opus-codec.org/docs/ */
-
-#include <stdlib.h>
-#include <errno.h>
-#include <string.h>
-#include <opus.h>
-#include <stdio.h>
-
-/*The frame size is hardcoded for this sample code but it doesn't have to be*/
-#define FRAME_SIZE 960
-#define SAMPLE_RATE 48000
-#define CHANNELS 2
-#define APPLICATION OPUS_APPLICATION_AUDIO
-#define BITRATE 64000
-
-#define MAX_FRAME_SIZE 6*960
-#define MAX_PACKET_SIZE (3*1276)
-
-int main(int argc, char **argv)
-{
-   char *inFile;
-   FILE *fin;
-   char *outFile;
-   FILE *fout;
-   opus_int16 in[FRAME_SIZE*CHANNELS];
-   opus_int16 out[MAX_FRAME_SIZE*CHANNELS];
-   unsigned char cbits[MAX_PACKET_SIZE];
-   int nbBytes;
-   /*Holds the state of the encoder and decoder */
-   OpusEncoder *encoder;
-   OpusDecoder *decoder;
-   int err;
-
-   if (argc != 3)
-   {
-      fprintf(stderr, "usage: trivial_example input.pcm output.pcm\n");
-      fprintf(stderr, "input and output are 16-bit little-endian raw files\n");
-      return EXIT_FAILURE;
-   }
-
-   /*Create a new encoder state */
-   encoder = opus_encoder_create(SAMPLE_RATE, CHANNELS, APPLICATION, &err);
-   if (err<0)
-   {
-      fprintf(stderr, "failed to create an encoder: %s\n", opus_strerror(err));
-      return EXIT_FAILURE;
-   }
-   /* Set the desired bit-rate. You can also set other parameters if needed.
-      The Opus library is designed to have good defaults, so only set
-      parameters you know you need. Doing otherwise is likely to result
-      in worse quality, but better. */
-   err = opus_encoder_ctl(encoder, OPUS_SET_BITRATE(BITRATE));
-   if (err<0)
-   {
-      fprintf(stderr, "failed to set bitrate: %s\n", opus_strerror(err));
-      return EXIT_FAILURE;
-   }
-   inFile = argv[1];
-   fin = fopen(inFile, "rb");
-   if (fin==NULL)
-   {
-      fprintf(stderr, "failed to open input file: %s\n", strerror(errno));
-      return EXIT_FAILURE;
-   }
-
-
-   /* Create a new decoder state. */
-   decoder = opus_decoder_create(SAMPLE_RATE, CHANNELS, &err);
-   if (err<0)
-   {
-      fprintf(stderr, "failed to create decoder: %s\n", opus_strerror(err));
-      return EXIT_FAILURE;
-   }
-   outFile = argv[2];
-   fout = fopen(outFile, "wb");
-   if (fout==NULL)
-   {
-      fprintf(stderr, "failed to open output file: %s\n", strerror(errno));
-      return EXIT_FAILURE;
-   }
-
-   while (1)
-   {
-      int i;
-      unsigned char pcm_bytes[MAX_FRAME_SIZE*CHANNELS*2];
-      int frame_size;
-      size_t samples;
-
-      /* Read a 16 bits/sample audio frame. */
-      samples = fread(pcm_bytes, sizeof(short)*CHANNELS, FRAME_SIZE, fin);
-
-      /* For simplicity, only read whole frames. In a real application,
-       * we'd pad the final partial frame with zeroes, record the exact
-       * duration, and trim the decoded audio to match.
-       */
-      if (samples != FRAME_SIZE)
-      {
-         break;
-      }
-
-      /* Convert from little-endian ordering. */
-      for (i=0;i<CHANNELS*FRAME_SIZE;i++)
-      {
-         in[i]=pcm_bytes[2*i+1]<<8|pcm_bytes[2*i];
-      }
-
-      /* Encode the frame. */
-      nbBytes = opus_encode(encoder, in, FRAME_SIZE, cbits, MAX_PACKET_SIZE);
-      if (nbBytes<0)
-      {
-         fprintf(stderr, "encode failed: %s\n", opus_strerror(nbBytes));
-         return EXIT_FAILURE;
-      }
-
-
-      /* Decode the data. In this example, frame_size will be constant because
-         the encoder is using a constant frame size. However, that may not
-         be the case for all encoders, so the decoder must always check
-         the frame size returned. */
-      frame_size = opus_decode(decoder, cbits, nbBytes, out, MAX_FRAME_SIZE, 0);
-      if (frame_size<0)
-      {
-         fprintf(stderr, "decoder failed: %s\n", opus_strerror(frame_size));
-         return EXIT_FAILURE;
-      }
-
-      /* Convert to little-endian ordering. */
-      for(i=0;i<CHANNELS*frame_size;i++)
-      {
-         pcm_bytes[2*i]=out[i]&0xFF;
-         pcm_bytes[2*i+1]=(out[i]>>8)&0xFF;
-      }
-      /* Write the decoded audio to file. */
-      fwrite(pcm_bytes, sizeof(short), frame_size*CHANNELS, fout);
-   }
-   /*Destroy the encoder state*/
-   opus_encoder_destroy(encoder);
-   opus_decoder_destroy(decoder);
-   fclose(fin);
-   fclose(fout);
-   return EXIT_SUCCESS;
-}

package/opus/include/meson.build

@@ -1,13 +0,0 @@
-opus_headers = [
-  'opus.h',
-  'opus_multistream.h',
-  'opus_projection.h',
-  'opus_types.h',
-  'opus_defines.h',
-]
-
-if opt_custom_modes
-  opus_headers += ['opus_custom.h']
-endif
-
-install_headers(opus_headers, subdir: 'opus')