opus: Update to upstream libopus 1.1.3

thirdparty/README.md

@@ -141,7 +141,7 @@ changes are marked with `// -- GODOT --` comments.
 ## openssl
 
 - Upstream: https://www.openssl.org
-- Version: 1.2.0h
+- Version: 1.0.2h
 - License: OpenSSL license / BSD-like
 
 Files extracted from the upstream source:
@@ -152,7 +152,7 @@ TODO.
 ## opus
 
 - Upstream: https://opus-codec.org
-- Version: 1.1.2 (opus) and 0.7 (opusfile)
+- Version: 1.1.3 (opus) and 0.7 (opusfile)
 - License: BSD-3-Clause
 
 Files extracted from upstream source:

thirdparty/opus/COPYING

-Copyright (c) 1994-2013 Xiph.Org Foundation and contributors
+Copyright 2001-2011 Xiph.Org, Skype Limited, Octasic,
+                    Jean-Marc Valin, Timothy B. Terriberry,
+                    CSIRO, Gregory Maxwell, Mark Borgerding,
+                    Erik de Castro Lopo
 
 Redistribution and use in source and binary forms, with or without
 modification, are permitted provided that the following conditions
@@ -11,18 +14,31 @@ notice, this list of conditions and the following disclaimer.
 notice, this list of conditions and the following disclaimer in the
 documentation and/or other materials provided with the distribution.
 
-- Neither the name of the Xiph.Org Foundation nor the names of its
-contributors may be used to endorse or promote products derived from
-this software without specific prior written permission.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
 
 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 FOUNDATION
-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.
+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.
+
+Opus is subject to the royalty-free patent licenses which are
+specified at:
+
+Xiph.Org Foundation:
+https://datatracker.ietf.org/ipr/1524/
+
+Microsoft Corporation:
+https://datatracker.ietf.org/ipr/1914/
+
+Broadcom Corporation:
+https://datatracker.ietf.org/ipr/1526/

thirdparty/opus/analysis.c

@@ -540,17 +540,14 @@ static void tonality_analysis(TonalityAnalysisState *tonal, const CELTMode *celt
        /* Instantaneous probability of speech and music, with beta pre-applied. */
        float speech0;
        float music0;
+       float p, q;
 
        /* One transition every 3 minutes of active audio */
        tau = .00005f*frame_probs[1];
-       beta = .05f;
-       if (1) {
-          /* Adapt beta based on how "unexpected" the new prob is */
-          float p, q;
-          p = MAX16(.05f,MIN16(.95f,frame_probs[0]));
-          q = MAX16(.05f,MIN16(.95f,tonal->music_prob));
-          beta = .01f+.05f*ABS16(p-q)/(p*(1-q)+q*(1-p));
-       }
+       /* Adapt beta based on how "unexpected" the new prob is */
+       p = MAX16(.05f,MIN16(.95f,frame_probs[0]));
+       q = MAX16(.05f,MIN16(.95f,tonal->music_prob));
+       beta = .01f+.05f*ABS16(p-q)/(p*(1-q)+q*(1-p));
        /* p0 and p1 are the probabilities of speech and music at this frame
           using only information from previous frame and applying the
           state transition model */

thirdparty/opus/opus.c

@@ -104,6 +104,10 @@ OPUS_EXPORT void opus_pcm_soft_clip(float *_x, int N, int C, float *declip_mem)
 
          /* Compute a such that maxval + a*maxval^2 = 1 */
          a=(maxval-1)/(maxval*maxval);
+         /* Slightly boost "a" by 2^-22. This is just enough to ensure -ffast-math
+            does not cause output values larger than +/-1, but small enough not
+            to matter even for 24-bit output.  */
+         a += a*2.4e-7;
          if (x[i*C]>0)
             a = -a;
          /* Apply soft clipping */
@@ -201,8 +205,10 @@ int opus_packet_parse_impl(const unsigned char *data, opus_int32 len,
    opus_int32 pad = 0;
    const unsigned char *data0 = data;
 
-   if (size==NULL)
+   if (size==NULL || len<0)
       return OPUS_BAD_ARG;
+   if (len==0)
+      return OPUS_INVALID_PACKET;
 
    framesize = opus_packet_get_samples_per_frame(data, 48000);
 

thirdparty/opus/opus/opus.h

@@ -142,7 +142,7 @@ extern "C" {
   *
   * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
   * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
-  * is 1 byte, then the packet does not need to be transmitted (DTX).
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
   *
   * Once the encoder state if no longer needed, it can be destroyed with
   *

thirdparty/opus/opus/opus_defines.h

@@ -65,7 +65,7 @@ extern "C" {
 
 #ifndef OPUS_EXPORT
 # if defined(WIN32)
-#  ifdef OPUS_BUILD
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
 #   define OPUS_EXPORT __declspec(dllexport)
 #  else
 #   define OPUS_EXPORT

thirdparty/opus/opus/opus_multistream.h

@@ -110,10 +110,10 @@ extern "C" {
   * packets produced by the encoder. Some basic information, such as packet
   * duration, can be computed without any special negotiation.
   *
-  * The format for multistream Opus packets is defined in the
-  * <a href="https://tools.ietf.org/html/draft-ietf-codec-oggopus">Ogg
-  * encapsulation specification</a> and is based on the self-delimited Opus
-  * framing described in Appendix B of <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
   * Normal Opus packets are just a degenerate case of multistream Opus packets,
   * and can be encoded or decoded with the multistream API by setting
   * <code>streams</code> to <code>1</code> when initializing the encoder or

thirdparty/opus/opus_encoder.c

@@ -860,9 +860,6 @@ opus_int32 compute_frame_size(const void *analysis_pcm, int frame_size,
 
 opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int32 Fs, StereoWidthState *mem)
 {
-   opus_val16 corr;
-   opus_val16 ldiff;
-   opus_val16 width;
    opus_val32 xx, xy, yy;
    opus_val16 sqrt_xx, sqrt_yy;
    opus_val16 qrrt_xx, qrrt_yy;
@@ -871,9 +868,12 @@ opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int3
    opus_val16 short_alpha;
 
    frame_rate = Fs/frame_size;
-   short_alpha = Q15ONE - 25*Q15ONE/IMAX(50,frame_rate);
+   short_alpha = Q15ONE - MULT16_16(25, Q15ONE)/IMAX(50,frame_rate);
    xx=xy=yy=0;
-   for (i=0;i<frame_size;i+=4)
+   /* Unroll by 4. The frame size is always a multiple of 4 *except* for
+      2.5 ms frames at 12 kHz. Since this setting is very rare (and very
+      stupid), we just discard the last two samples. */
+   for (i=0;i<frame_size-3;i+=4)
    {
       opus_val32 pxx=0;
       opus_val32 pxy=0;
@@ -912,6 +912,9 @@ opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int3
    mem->YY = MAX32(0, mem->YY);
    if (MAX32(mem->XX, mem->YY)>QCONST16(8e-4f, 18))
    {
+      opus_val16 corr;
+      opus_val16 ldiff;
+      opus_val16 width;
       sqrt_xx = celt_sqrt(mem->XX);
       sqrt_yy = celt_sqrt(mem->YY);
       qrrt_xx = celt_sqrt(sqrt_xx);
@@ -920,19 +923,15 @@ opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int3
       mem->XY = MIN32(mem->XY, sqrt_xx*sqrt_yy);
       corr = SHR32(frac_div32(mem->XY,EPSILON+MULT16_16(sqrt_xx,sqrt_yy)),16);
       /* Approximate loudness difference */
-      ldiff = Q15ONE*ABS16(qrrt_xx-qrrt_yy)/(EPSILON+qrrt_xx+qrrt_yy);
+      ldiff = MULT16_16(Q15ONE, ABS16(qrrt_xx-qrrt_yy))/(EPSILON+qrrt_xx+qrrt_yy);
       width = MULT16_16_Q15(celt_sqrt(QCONST32(1.f,30)-MULT16_16(corr,corr)), ldiff);
       /* Smoothing over one second */
       mem->smoothed_width += (width-mem->smoothed_width)/frame_rate;
       /* Peak follower */
       mem->max_follower = MAX16(mem->max_follower-QCONST16(.02f,15)/frame_rate, mem->smoothed_width);
-   } else {
-      width = 0;
-      corr=Q15ONE;
-      ldiff=0;
    }
    /*printf("%f %f %f %f %f ", corr/(float)Q15ONE, ldiff/(float)Q15ONE, width/(float)Q15ONE, mem->smoothed_width/(float)Q15ONE, mem->max_follower/(float)Q15ONE);*/
-   return EXTRACT16(MIN32(Q15ONE,20*mem->max_follower));
+   return EXTRACT16(MIN32(Q15ONE, MULT16_16(20, mem->max_follower)));
 }
 
 opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
@@ -1050,6 +1049,16 @@ opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_
     st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes);
 
     frame_rate = st->Fs/frame_size;
+    if (!st->use_vbr)
+    {
+       int cbrBytes;
+       /* Multiply by 3 to make sure the division is exact. */
+       int frame_rate3 = 3*st->Fs/frame_size;
+       /* We need to make sure that "int" values always fit in 16 bits. */
+       cbrBytes = IMIN( (3*st->bitrate_bps/8 + frame_rate3/2)/frame_rate3, max_data_bytes);
+       st->bitrate_bps = cbrBytes*(opus_int32)frame_rate3*8/3;
+       max_data_bytes = cbrBytes;
+    }
     if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8
        || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400)))
     {
@@ -1066,18 +1075,18 @@ opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_
           bw=OPUS_BANDWIDTH_WIDEBAND;
        else if (tocmode==MODE_CELT_ONLY&&bw==OPUS_BANDWIDTH_MEDIUMBAND)
           bw=OPUS_BANDWIDTH_NARROWBAND;
-       else if (bw<=OPUS_BANDWIDTH_SUPERWIDEBAND)
+       else if (tocmode==MODE_HYBRID&&bw<=OPUS_BANDWIDTH_SUPERWIDEBAND)
           bw=OPUS_BANDWIDTH_SUPERWIDEBAND;
        data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels);
+       ret = 1;
+       if (!st->use_vbr)
+       {
+          ret = opus_packet_pad(data, ret, max_data_bytes);
+          if (ret == OPUS_OK)
+             ret = max_data_bytes;
+       }
        RESTORE_STACK;
-       return 1;
-    }
-    if (!st->use_vbr)
-    {
-       int cbrBytes;
-       cbrBytes = IMIN( (st->bitrate_bps + 4*frame_rate)/(8*frame_rate) , max_data_bytes);
-       st->bitrate_bps = cbrBytes * (8*frame_rate);
-       max_data_bytes = cbrBytes;
+       return ret;
     }
     max_rate = frame_rate*max_data_bytes*8;
 
@@ -1513,7 +1522,7 @@ opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_
                celt_rate = total_bitRate - st->silk_mode.bitRate;
                HB_gain_ref = (curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND) ? 3000 : 3600;
                HB_gain = SHL32((opus_val32)celt_rate, 9) / SHR32((opus_val32)celt_rate + st->stream_channels * HB_gain_ref, 6);
-               HB_gain = HB_gain < Q15ONE*6/7 ? HB_gain + Q15ONE/7 : Q15ONE;
+               HB_gain = HB_gain < (opus_val32)Q15ONE*6/7 ? HB_gain + Q15ONE/7 : Q15ONE;
             }
         } else {
             /* SILK gets all bits */

thirdparty/opus/repacketizer.c

@@ -249,7 +249,9 @@ int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len)
    opus_repacketizer_init(&rp);
    /* Moving payload to the end of the packet so we can do in-place padding */
    OPUS_MOVE(data+new_len-len, data, len);
-   opus_repacketizer_cat(&rp, data+new_len-len, len);
+   ret = opus_repacketizer_cat(&rp, data+new_len-len, len);
+   if (ret != OPUS_OK)
+      return ret;
    ret = opus_repacketizer_out_range_impl(&rp, 0, rp.nb_frames, data, new_len, 0, 1);
    if (ret > 0)
       return OPUS_OK;