@scrypted/libav 1.0.166 → 1.0.168

package/opus/doc/opus_logo.svg

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

@@ -0,0 +1,244 @@
+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

@@ -0,0 +1,43 @@
+= 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

@@ -0,0 +1,171 @@
+/* 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

@@ -0,0 +1,13 @@
+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')