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

@@ -1,4 +1,7 @@
-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
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
-contributors may be used to endorse or promote products derived from
+names of specific contributors, may be used to endorse or promote
-this software without specific prior written permission.
+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
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
-OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+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;
+       /* Adapt beta based on how "unexpected" the new prob is */
-       if (1) {
+       p = MAX16(.05f,MIN16(.95f,frame_probs[0]));
-          /* Adapt beta based on how "unexpected" the new prob is */
+       q = MAX16(.05f,MIN16(.95f,tonal->music_prob));
-          float p, q;
+       beta = .01f+.05f*ABS16(p-q)/(p*(1-q)+q*(1-p));
-          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
+  * The format for multistream Opus packets is defined in
-  * <a href="https://tools.ietf.org/html/draft-ietf-codec-oggopus">Ogg
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
-  * encapsulation specification</a> and is based on the self-delimited Opus
+  * and is based on the self-delimited Opus framing described in Appendix B of
-  * framing described in Appendix B of <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * <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;
+       return ret;
-    }
-    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;
     }
     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/opus_multistream_encoder.c

@@ -70,13 +70,22 @@ typedef void (*opus_copy_channel_in_func)(
   int frame_size
 );
 
+typedef enum {
+  MAPPING_TYPE_NONE,
+  MAPPING_TYPE_SURROUND
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+  ,  /* Do not include comma at end of enumerator list */
+  MAPPING_TYPE_AMBISONICS
+#endif
+} MappingType;
+
 struct OpusMSEncoder {
    ChannelLayout layout;
    int arch;
    int lfe_stream;
    int application;
    int variable_duration;
-   int surround;
+   MappingType mapping_type;
    opus_int32 bitrate_bps;
    float subframe_mem[3];
    /* Encoder states go here */
@@ -242,6 +251,7 @@ void surround_analysis(const CELTMode *celt_mode, const void *pcm, opus_val16 *b
    upsample = resampling_factor(rate);
    frame_size = len*upsample;
 
+   /* LM = log2(frame_size / 120) */
    for (LM=0;LM<celt_mode->maxLM;LM++)
       if (celt_mode->shortMdctSize<<LM==frame_size)
          break;
@@ -398,6 +408,12 @@ opus_int32 opus_multistream_surround_encoder_get_size(int channels, int mapping_
    {
       nb_streams=channels;
       nb_coupled_streams=0;
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+   } else if (mapping_family==254)
+   {
+      nb_streams=channels;
+      nb_coupled_streams=0;
+#endif
    } else
       return 0;
    size = opus_multistream_encoder_get_size(nb_streams, nb_coupled_streams);
@@ -408,7 +424,6 @@ opus_int32 opus_multistream_surround_encoder_get_size(int channels, int mapping_
    return size;
 }
 
-
 static int opus_multistream_encoder_init_impl(
       OpusMSEncoder *st,
       opus_int32 Fs,
@@ -417,7 +432,7 @@ static int opus_multistream_encoder_init_impl(
       int coupled_streams,
       const unsigned char *mapping,
       int application,
-      int surround
+      MappingType mapping_type
 )
 {
    int coupled_size;
@@ -434,7 +449,7 @@ static int opus_multistream_encoder_init_impl(
    st->layout.nb_streams = streams;
    st->layout.nb_coupled_streams = coupled_streams;
    st->subframe_mem[0]=st->subframe_mem[1]=st->subframe_mem[2]=0;
-   if (!surround)
+   if (mapping_type != MAPPING_TYPE_SURROUND)
       st->lfe_stream = -1;
    st->bitrate_bps = OPUS_AUTO;
    st->application = application;
@@ -463,12 +478,12 @@ static int opus_multistream_encoder_init_impl(
       if(ret!=OPUS_OK)return ret;
       ptr += align(mono_size);
    }
-   if (surround)
+   if (mapping_type == MAPPING_TYPE_SURROUND)
    {
       OPUS_CLEAR(ms_get_preemph_mem(st), channels);
       OPUS_CLEAR(ms_get_window_mem(st), channels*120);
    }
-   st->surround = surround;
+   st->mapping_type = mapping_type;
    return OPUS_OK;
 }
 
@@ -482,7 +497,9 @@ int opus_multistream_encoder_init(
       int application
 )
 {
-   return opus_multistream_encoder_init_impl(st, Fs, channels, streams, coupled_streams, mapping, application, 0);
+   return opus_multistream_encoder_init_impl(st, Fs, channels, streams,
+                                             coupled_streams, mapping,
+                                             application, MAPPING_TYPE_NONE);
 }
 
 int opus_multistream_surround_encoder_init(
@@ -496,6 +513,8 @@ int opus_multistream_surround_encoder_init(
       int application
 )
 {
+   MappingType mapping_type;
+
    if ((channels>255) || (channels<1))
       return OPUS_BAD_ARG;
    st->lfe_stream = -1;
@@ -530,10 +549,32 @@ int opus_multistream_surround_encoder_init(
       *coupled_streams=0;
       for(i=0;i<channels;i++)
          mapping[i] = i;
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+   } else if (mapping_family==254)
+   {
+      int i;
+      *streams=channels;
+      *coupled_streams=0;
+      for(i=0;i<channels;i++)
+         mapping[i] = i;
+#endif
    } else
       return OPUS_UNIMPLEMENTED;
-   return opus_multistream_encoder_init_impl(st, Fs, channels, *streams, *coupled_streams,
+
-         mapping, application, channels>2&&mapping_family==1);
+   if (channels>2 && mapping_family==1) {
+      mapping_type = MAPPING_TYPE_SURROUND;
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+   } else if (mapping_family==254)
+   {
+      mapping_type = MAPPING_TYPE_AMBISONICS;
+#endif
+   } else
+   {
+      mapping_type = MAPPING_TYPE_NONE;
+   }
+   return opus_multistream_encoder_init_impl(st, Fs, channels, *streams,
+                                             *coupled_streams, mapping,
+                                             application, mapping_type);
 }
 
 OpusMSEncoder *opus_multistream_encoder_create(
@@ -618,24 +659,19 @@ OpusMSEncoder *opus_multistream_surround_encoder_create(
    return st;
 }
 
-static opus_int32 surround_rate_allocation(
+static void surround_rate_allocation(
       OpusMSEncoder *st,
       opus_int32 *rate,
-      int frame_size
+      int frame_size,
+      opus_int32 Fs
       )
 {
    int i;
    opus_int32 channel_rate;
-   opus_int32 Fs;
-   char *ptr;
    int stream_offset;
    int lfe_offset;
    int coupled_ratio; /* Q8 */
    int lfe_ratio;     /* Q8 */
-   opus_int32 rate_sum=0;
-
-   ptr = (char*)st + align(sizeof(OpusMSEncoder));
-   opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs));
 
    if (st->bitrate_bps > st->layout.nb_channels*40000)
       stream_offset = 20000;
@@ -688,6 +724,88 @@ static opus_int32 surround_rate_allocation(
          rate[i] = stream_offset+channel_rate;
       else
          rate[i] = lfe_offset+(channel_rate*lfe_ratio>>8);
+   }
+}
+
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+static void ambisonics_rate_allocation(
+      OpusMSEncoder *st,
+      opus_int32 *rate,
+      int frame_size,
+      opus_int32 Fs
+      )
+{
+   int i;
+   int non_mono_rate;
+   int total_rate;
+
+   /* The mono channel gets (rate_ratio_num / rate_ratio_den) times as many bits
+    * as all other channels */
+   const int rate_ratio_num = 4;
+   const int rate_ratio_den = 3;
+   const int num_channels = st->layout.nb_streams;
+
+   if (st->bitrate_bps==OPUS_AUTO)
+   {
+      total_rate = num_channels * (20000 + st->layout.nb_streams*(Fs+60*Fs/frame_size));
+   } else if (st->bitrate_bps==OPUS_BITRATE_MAX)
+   {
+      total_rate = num_channels * 320000;
+   } else {
+      total_rate = st->bitrate_bps;
+   }
+
+   /* Let y be the non-mono rate and let p, q be integers such that the mono
+    * channel rate is (p/q) * y.
+    * Also let T be the total bitrate to allocate. Then
+    *   (n - 1) y + (p/q) y = T
+    *   y = (T q) / (qn - q + p)
+    */
+   non_mono_rate =
+         total_rate * rate_ratio_den
+         / (rate_ratio_den*num_channels + rate_ratio_num - rate_ratio_den);
+
+#ifndef FIXED_POINT
+   if (st->variable_duration==OPUS_FRAMESIZE_VARIABLE && frame_size != Fs/50)
+   {
+      opus_int32 bonus = 60*(Fs/frame_size-50);
+      non_mono_rate += bonus;
+   }
+#endif
+
+   rate[0] = total_rate - (num_channels - 1) * non_mono_rate;
+   for (i=1;i<st->layout.nb_streams;i++)
+   {
+      rate[i] = non_mono_rate;
+   }
+}
+#endif /* ENABLE_EXPERIMENTAL_AMBISONICS */
+
+static opus_int32 rate_allocation(
+      OpusMSEncoder *st,
+      opus_int32 *rate,
+      int frame_size
+      )
+{
+   int i;
+   opus_int32 rate_sum=0;
+   opus_int32 Fs;
+   char *ptr;
+
+   ptr = (char*)st + align(sizeof(OpusMSEncoder));
+   opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs));
+
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+   if (st->mapping_type == MAPPING_TYPE_AMBISONICS) {
+     ambisonics_rate_allocation(st, rate, frame_size, Fs);
+   } else
+#endif
+   {
+     surround_rate_allocation(st, rate, frame_size, Fs);
+   }
+
+   for (i=0;i<st->layout.nb_streams;i++)
+   {
       rate[i] = IMAX(rate[i], 500);
       rate_sum += rate[i];
    }
@@ -730,7 +848,7 @@ static int opus_multistream_encode_native
    opus_int32 smallest_packet;
    ALLOC_STACK;
 
-   if (st->surround)
+   if (st->mapping_type == MAPPING_TYPE_SURROUND)
    {
       preemph_mem = ms_get_preemph_mem(st);
       mem = ms_get_window_mem(st);
@@ -784,13 +902,13 @@ static int opus_multistream_encode_native
    mono_size = opus_encoder_get_size(1);
 
    ALLOC(bandSMR, 21*st->layout.nb_channels, opus_val16);
-   if (st->surround)
+   if (st->mapping_type == MAPPING_TYPE_SURROUND)
    {
       surround_analysis(celt_mode, pcm, bandSMR, mem, preemph_mem, frame_size, 120, st->layout.nb_channels, Fs, copy_channel_in, st->arch);
    }
 
    /* Compute bitrate allocation between streams (this could be a lot better) */
-   rate_sum = surround_rate_allocation(st, bitrates, frame_size);
+   rate_sum = rate_allocation(st, bitrates, frame_size);
 
    if (!vbr)
    {
@@ -813,7 +931,7 @@ static int opus_multistream_encode_native
       else
          ptr += align(mono_size);
       opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrates[s]));
-      if (st->surround)
+      if (st->mapping_type == MAPPING_TYPE_SURROUND)
       {
          opus_int32 equiv_rate;
          equiv_rate = st->bitrate_bps;
@@ -834,6 +952,11 @@ static int opus_multistream_encode_native
             opus_encoder_ctl(enc, OPUS_SET_FORCE_CHANNELS(2));
          }
       }
+#ifdef ENABLE_EXPERIMENTAL_AMBISONICS
+      else if (st->mapping_type == MAPPING_TYPE_AMBISONICS) {
+        opus_encoder_ctl(enc, OPUS_SET_FORCE_MODE(MODE_CELT_ONLY));
+      }
+#endif
    }
 
    ptr = (char*)st + align(sizeof(OpusMSEncoder));
@@ -845,6 +968,7 @@ static int opus_multistream_encode_native
       int len;
       int curr_max;
       int c1, c2;
+      int ret;
 
       opus_repacketizer_init(&rp);
       enc = (OpusEncoder*)ptr;
@@ -859,7 +983,7 @@ static int opus_multistream_encode_native
          (*copy_channel_in)(buf+1, 2,
             pcm, st->layout.nb_channels, right, frame_size);
          ptr += align(coupled_size);
-         if (st->surround)
+         if (st->mapping_type == MAPPING_TYPE_SURROUND)
          {
             for (i=0;i<21;i++)
             {
@@ -875,7 +999,7 @@ static int opus_multistream_encode_native
          (*copy_channel_in)(buf, 1,
             pcm, st->layout.nb_channels, chan, frame_size);
          ptr += align(mono_size);
-         if (st->surround)
+         if (st->mapping_type == MAPPING_TYPE_SURROUND)
          {
             for (i=0;i<21;i++)
                bandLogE[i] = bandSMR[21*chan+i];
@@ -883,7 +1007,7 @@ static int opus_multistream_encode_native
          c1 = chan;
          c2 = -1;
       }
-      if (st->surround)
+      if (st->mapping_type == MAPPING_TYPE_SURROUND)
          opus_encoder_ctl(enc, OPUS_SET_ENERGY_MASK(bandLogE));
       /* number of bytes left (+Toc) */
       curr_max = max_data_bytes - tot_size;
@@ -904,7 +1028,14 @@ static int opus_multistream_encode_native
       /* We need to use the repacketizer to add the self-delimiting lengths
          while taking into account the fact that the encoder can now return
          more than one frame at a time (e.g. 60 ms CELT-only) */
-      opus_repacketizer_cat(&rp, tmp_data, len);
+      ret = opus_repacketizer_cat(&rp, tmp_data, len);
+      /* If the opus_repacketizer_cat() fails, then something's seriously wrong
+         with the encoder. */
+      if (ret != OPUS_OK)
+      {
+         RESTORE_STACK;
+         return OPUS_INTERNAL_ERROR;
+      }
       len = opus_repacketizer_out_range_impl(&rp, 0, opus_repacketizer_get_nb_frames(&rp),
             data, max_data_bytes-tot_size, s != st->layout.nb_streams-1, !vbr && s == st->layout.nb_streams-1);
       data += len;
@@ -1183,7 +1314,7 @@ int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...)
    {
       int s;
       st->subframe_mem[0] = st->subframe_mem[1] = st->subframe_mem[2] = 0;
-      if (st->surround)
+      if (st->mapping_type == MAPPING_TYPE_SURROUND)
       {
          OPUS_CLEAR(ms_get_preemph_mem(st), st->layout.nb_channels);
          OPUS_CLEAR(ms_get_window_mem(st), st->layout.nb_channels*120);

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;