int test_decoder_code0(int no_fuzz)
{
static const opus_int32 fsv[5]={48000,24000,16000,12000,8000};
int err,skip,plen;
int out_samples,fec;
int t;
opus_int32 i;
OpusDecoder *dec[5*2];
opus_int32 decsize;
OpusDecoder *decbak;
opus_uint32 dec_final_range1,dec_final_range2,dec_final_acc;
unsigned char *packet;
unsigned char modes[4096];
short *outbuf_int;
short *outbuf;
dec_final_range1=dec_final_range2=2;
packet=malloc(sizeof(unsigned char)*MAX_PACKET);
if(packet==NULL)test_failed();
outbuf_int=malloc(sizeof(short)*(MAX_FRAME_SAMP+16)*2);
for(i=0;i<(MAX_FRAME_SAMP+16)*2;i++)outbuf_int[i]=32749;
outbuf=&outbuf_int[8*2];
fprintf(stdout," Starting %d decoders...\n",5*2);
for(t=0;t<5*2;t++)
{
int fs=fsv[t>>1];
int c=(t&1)+1;
err=OPUS_INTERNAL_ERROR;
dec[t] = opus_decoder_create(fs, c, &err);
if(err!=OPUS_OK || dec[t]==NULL)test_failed();
fprintf(stdout," opus_decoder_create(%5d,%d) OK. Copy ",fs,c);
{
OpusDecoder *dec2;
/*The opus state structures contain no pointers and can be freely copied*/
dec2=(OpusDecoder *)malloc(opus_decoder_get_size(c));
if(dec2==NULL)test_failed();
memcpy(dec2,dec[t],opus_decoder_get_size(c));
memset(dec[t],255,opus_decoder_get_size(c));
opus_decoder_destroy(dec[t]);
printf("OK.\n");
dec[t]=dec2;
}
}
decsize=opus_decoder_get_size(1);
decbak=(OpusDecoder *)malloc(decsize);
if(decbak==NULL)test_failed();
for(t=0;t<5*2;t++)
{
int factor=48000/fsv[t>>1];
for(fec=0;fec<2;fec++)
{
opus_int32 dur;
/*Test PLC on a fresh decoder*/
out_samples = opus_decode(dec[t], 0, 0, outbuf, 120/factor, fec);
if(out_samples!=120/factor)test_failed();
if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed();
if(dur!=120/factor)test_failed();
/*Test on a size which isn't a multiple of 2.5ms*/
out_samples = opus_decode(dec[t], 0, 0, outbuf, 120/factor+2, fec);
if(out_samples!=OPUS_BAD_ARG)test_failed();
/*Test null pointer input*/
out_samples = opus_decode(dec[t], 0, -1, outbuf, 120/factor, fec);
if(out_samples!=120/factor)test_failed();
out_samples = opus_decode(dec[t], 0, 1, outbuf, 120/factor, fec);
if(out_samples!=120/factor)test_failed();
out_samples = opus_decode(dec[t], 0, 10, outbuf, 120/factor, fec);
if(out_samples!=120/factor)test_failed();
out_samples = opus_decode(dec[t], 0, fast_rand(), outbuf, 120/factor, fec);
if(out_samples!=120/factor)test_failed();
if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed();
if(dur!=120/factor)test_failed();
/*Zero lengths*/
out_samples = opus_decode(dec[t], packet, 0, outbuf, 120/factor, fec);
if(out_samples!=120/factor)test_failed();
/*Zero buffer*/
outbuf[0]=32749;
out_samples = opus_decode(dec[t], packet, 0, outbuf, 0, fec);
if(out_samples>0)test_failed();
#if !defined(OPUS_BUILD) && (OPUS_GNUC_PREREQ(4, 6) || (defined(__clang_major__) && __clang_major__ >= 3))
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wnonnull"
#endif
out_samples = opus_decode(dec[t], packet, 0, 0, 0, fec);
#if !defined(OPUS_BUILD) && (OPUS_GNUC_PREREQ(4, 6) || (defined(__clang_major__) && __clang_major__ >= 3))
#pragma GCC diagnostic pop
#endif
if(out_samples>0)test_failed();
if(outbuf[0]!=32749)test_failed();
/*Invalid lengths*/
out_samples = opus_decode(dec[t], packet, -1, outbuf, MAX_FRAME_SAMP, fec);
if(out_samples>=0)test_failed();
out_samples = opus_decode(dec[t], packet, INT_MIN, outbuf, MAX_FRAME_SAMP, fec);
if(out_samples>=0)test_failed();
out_samples = opus_decode(dec[t], packet, -1, outbuf, -1, fec);
if(out_samples>=0)test_failed();
/*Crazy FEC values*/
out_samples = opus_decode(dec[t], packet, 1, outbuf, MAX_FRAME_SAMP, fec?-1:2);
if(out_samples>=0)test_failed();
/*Reset the decoder*/
if(opus_decoder_ctl(dec[t], OPUS_RESET_STATE)!=OPUS_OK)test_failed();
}
}
fprintf(stdout," dec[all] initial frame PLC OK.\n");
/*Count code 0 tests*/
for(i=0;i<64;i++)
{
opus_int32 dur;
int j,expected[5*2];
packet[0]=i<<2;
packet[1]=255;
packet[2]=255;
err=opus_packet_get_nb_channels(packet);
if(err!=(i&1)+1)test_failed();
for(t=0;t<5*2;t++){
expected[t]=opus_decoder_get_nb_samples(dec[t],packet,1);
if(expected[t]>2880)test_failed();
}
for(j=0;j<256;j++)
{
packet[1]=j;
for(t=0;t<5*2;t++)
{
out_samples = opus_decode(dec[t], packet, 3, outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=expected[t])test_failed();
if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed();
if(dur!=out_samples)test_failed();
opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1));
if(t==0)dec_final_range2=dec_final_range1;
else if(dec_final_range1!=dec_final_range2)test_failed();
}
}
for(t=0;t<5*2;t++){
int factor=48000/fsv[t>>1];
/* The PLC is run for 6 frames in order to get better PLC coverage. */
for(j=0;j<6;j++)
{
out_samples = opus_decode(dec[t], 0, 0, outbuf, expected[t], 0);
if(out_samples!=expected[t])test_failed();
if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed();
if(dur!=out_samples)test_failed();
}
/* Run the PLC once at 2.5ms, as a simulation of someone trying to
do small drift corrections. */
if(expected[t]!=120/factor)
{
out_samples = opus_decode(dec[t], 0, 0, outbuf, 120/factor, 0);
if(out_samples!=120/factor)test_failed();
if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed();
if(dur!=out_samples)test_failed();
}
out_samples = opus_decode(dec[t], packet, 2, outbuf, expected[t]-1, 0);
if(out_samples>0)test_failed();
}
}
fprintf(stdout," dec[all] all 2-byte prefix for length 3 and PLC, all modes (64) OK.\n");
if(no_fuzz)
{
fprintf(stdout," Skipping many tests which fuzz the decoder as requested.\n");
free(decbak);
for(t=0;t<5*2;t++)opus_decoder_destroy(dec[t]);
printf(" Decoders stopped.\n");
err=0;
for(i=0;i<8*2;i++)err|=outbuf_int[i]!=32749;
for(i=MAX_FRAME_SAMP*2;i<(MAX_FRAME_SAMP+8)*2;i++)err|=outbuf[i]!=32749;
if(err)test_failed();
free(outbuf_int);
free(packet);
return 0;
}
{
/*We only test a subset of the modes here simply because the longer
durations end up taking a long time.*/
static const int cmodes[4]={16,20,24,28};
static const opus_uint32 cres[4]={116290185,2172123586u,2172123586u,2172123586u};
static const opus_uint32 lres[3]={3285687739u,1481572662,694350475};
static const int lmodes[3]={0,4,8};
int mode=fast_rand()%4;
packet[0]=cmodes[mode]<<3;
dec_final_acc=0;
t=fast_rand()%10;
for(i=0;i<65536;i++)
{
int factor=48000/fsv[t>>1];
packet[1]=i>>8;
packet[2]=i&255;
packet[3]=255;
out_samples = opus_decode(dec[t], packet, 4, outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=120/factor)test_failed();
opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1));
dec_final_acc+=dec_final_range1;
}
if(dec_final_acc!=cres[mode])test_failed();
fprintf(stdout," dec[%3d] all 3-byte prefix for length 4, mode %2d OK.\n",t,cmodes[mode]);
mode=fast_rand()%3;
packet[0]=lmodes[mode]<<3;
dec_final_acc=0;
t=fast_rand()%10;
for(i=0;i<65536;i++)
{
int factor=48000/fsv[t>>1];
packet[1]=i>>8;
packet[2]=i&255;
packet[3]=255;
out_samples = opus_decode(dec[t], packet, 4, outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=480/factor)test_failed();
opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1));
dec_final_acc+=dec_final_range1;
}
if(dec_final_acc!=lres[mode])test_failed();
fprintf(stdout," dec[%3d] all 3-byte prefix for length 4, mode %2d OK.\n",t,lmodes[mode]);
}
skip=fast_rand()%7;
for(i=0;i<64;i++)
{
int j,expected[5*2];
packet[0]=i<<2;
for(t=0;t<5*2;t++)expected[t]=opus_decoder_get_nb_samples(dec[t],packet,1);
for(j=2+skip;j<1275;j+=4)
{
int jj;
for(jj=0;jj<j;jj++)packet[jj+1]=fast_rand()&255;
for(t=0;t<5*2;t++)
{
out_samples = opus_decode(dec[t], packet, j+1, outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=expected[t])test_failed();
opus_decoder_ctl(dec[t], OPUS_GET_FINAL_RANGE(&dec_final_range1));
if(t==0)dec_final_range2=dec_final_range1;
else if(dec_final_range1!=dec_final_range2)test_failed();
}
}
}
fprintf(stdout," dec[all] random packets, all modes (64), every 8th size from from %d bytes to maximum OK.\n",2+skip);
debruijn2(64,modes);
plen=(fast_rand()%18+3)*8+skip+3;
for(i=0;i<4096;i++)
{
int j,expected[5*2];
packet[0]=modes[i]<<2;
for(t=0;t<5*2;t++)expected[t]=opus_decoder_get_nb_samples(dec[t],packet,plen);
for(j=0;j<plen;j++)packet[j+1]=(fast_rand()|fast_rand())&255;
memcpy(decbak,dec[0],decsize);
if(opus_decode(decbak, packet, plen+1, outbuf, expected[0], 1)!=expected[0])test_failed();
memcpy(decbak,dec[0],decsize);
if(opus_decode(decbak, 0, 0, outbuf, MAX_FRAME_SAMP, 1)<20)test_failed();
memcpy(decbak,dec[0],decsize);
if(opus_decode(decbak, 0, 0, outbuf, MAX_FRAME_SAMP, 0)<20)test_failed();
for(t=0;t<5*2;t++)
{
opus_int32 dur;
out_samples = opus_decode(dec[t], packet, plen+1, outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=expected[t])test_failed();
if(t==0)dec_final_range2=dec_final_range1;
else if(dec_final_range1!=dec_final_range2)test_failed();
if(opus_decoder_ctl(dec[t], OPUS_GET_LAST_PACKET_DURATION(&dur))!=OPUS_OK)test_failed();
if(dur!=out_samples)test_failed();
}
}
fprintf(stdout," dec[all] random packets, all mode pairs (4096), %d bytes/frame OK.\n",plen+1);
plen=(fast_rand()%18+3)*8+skip+3;
t=rand()&3;
for(i=0;i<4096;i++)
{
int count,j,expected;
packet[0]=modes[i]<<2;
expected=opus_decoder_get_nb_samples(dec[t],packet,plen);
for(count=0;count<10;count++)
{
for(j=0;j<plen;j++)packet[j+1]=(fast_rand()|fast_rand())&255;
out_samples = opus_decode(dec[t], packet, plen+1, outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=expected)test_failed();
}
}
fprintf(stdout," dec[%3d] random packets, all mode pairs (4096)*10, %d bytes/frame OK.\n",t,plen+1);
{
int tmodes[1]={25<<2};
opus_uint32 tseeds[1]={140441};
int tlen[1]={157};
opus_int32 tret[1]={480};
t=fast_rand()&1;
for(i=0;i<1;i++)
{
int j;
packet[0]=tmodes[i];
Rw=Rz=tseeds[i];
for(j=1;j<tlen[i];j++)packet[j]=fast_rand()&255;
out_samples=opus_decode(dec[t], packet, tlen[i], outbuf, MAX_FRAME_SAMP, 0);
if(out_samples!=tret[i])test_failed();
}
fprintf(stdout," dec[%3d] pre-selected random packets OK.\n",t);
}
free(decbak);
for(t=0;t<5*2;t++)opus_decoder_destroy(dec[t]);
printf(" Decoders stopped.\n");
err=0;
for(i=0;i<8*2;i++)err|=outbuf_int[i]!=32749;
for(i=MAX_FRAME_SAMP*2;i<(MAX_FRAME_SAMP+8)*2;i++)err|=outbuf[i]!=32749;
if(err)test_failed();
free(outbuf_int);
free(packet);
return 0;
}
void FVoiceDecoderOpus::Decode(const uint8* InCompressedData, uint32 CompressedDataSize, uint8* OutRawPCMData, uint32& OutRawDataSize)
{
SCOPE_CYCLE_COUNTER(STAT_Voice_Decoding);
int32 HeaderSize = 0;
int32 BytesPerFrame = FrameSize * NumChannels * sizeof(opus_int16);
int32 MaxFramesEncoded = MAX_OPUS_UNCOMPRESSED_BUFFER_SIZE / BytesPerFrame;
int32 NumFramesToDecode = InCompressedData[0];
int32 PacketGeneration = InCompressedData[1];
HeaderSize += 2 * sizeof(uint8);
if (PacketGeneration != LastGeneration + 1)
{
UE_LOG(LogVoiceDecode, Verbose, TEXT("Packet generation skipped from %d to %d"), LastGeneration, PacketGeneration);
}
if (NumFramesToDecode > 0 && NumFramesToDecode <= MaxFramesEncoded)
{
// Start of compressed data offsets
const uint16* CompressedOffsets = (const uint16*)(InCompressedData + HeaderSize);
uint32 LengthOfCompressedOffsets = NumFramesToDecode * sizeof(uint16);
HeaderSize += LengthOfCompressedOffsets;
uint32 LengthOfEntropyOffsets = 0;
#if ADD_ENTROPY_TO_PACKET
// Start of the entropy to each encoded frame
const uint32* EntropyOffsets = (uint32*)(InCompressedData + HeaderSize);
LengthOfEntropyOffsets = NumFramesToDecode * sizeof(uint32);
HeaderSize += LengthOfEntropyOffsets;
#endif
// Start of compressed data
const uint8* CompressedDataStart = (InCompressedData + HeaderSize);
int32 CompressedBufferOffset = 0;
int32 DecompressedBufferOffset = 0;
uint16 LastCompressedOffset = 0;
for (int32 i=0; i < NumFramesToDecode; i++)
{
int32 UncompressedBufferAvail = OutRawDataSize - DecompressedBufferOffset;
if (UncompressedBufferAvail >= (MAX_OPUS_FRAMES * BytesPerFrame))
{
int32 CompressedBufferSize = CompressedOffsets[i] - LastCompressedOffset;
check(Decoder);
int32 NumDecompressedSamples = opus_decode(Decoder,
CompressedDataStart + CompressedBufferOffset, CompressedBufferSize,
(opus_int16*)(OutRawPCMData + DecompressedBufferOffset), MAX_OPUS_FRAME_SIZE, 0);
#if DEBUG_OPUS
DebugFrameDecodeInfo(CompressedDataStart + CompressedBufferOffset, CompressedBufferSize, SampleRate);
#endif // DEBUG_OPUS
if (NumDecompressedSamples < 0)
{
const char* ErrorStr = opus_strerror(NumDecompressedSamples);
UE_LOG(LogVoiceDecode, Warning, TEXT("Failed to decode: [%d] %s"), NumDecompressedSamples, ANSI_TO_TCHAR(ErrorStr));
break;
}
else
{
if (NumDecompressedSamples != FrameSize)
{
UE_LOG(LogVoiceDecode, Warning, TEXT("Unexpected decode result NumSamplesDecoded %d != FrameSize %d"), NumDecompressedSamples, FrameSize);
}
opus_decoder_ctl(Decoder, OPUS_GET_FINAL_RANGE(&Entropy[LastEntropyIdx]));
#if ADD_ENTROPY_TO_PACKET
if (Entropy[LastEntropyIdx] != EntropyOffsets[i])
{
UE_LOG(LogVoiceDecode, Verbose, TEXT("Decoder Entropy[%d/%d] = %d expected %d"), i, NumFramesToDecode-1, Entropy[LastEntropyIdx], EntropyOffsets[i]);
}
#endif
LastEntropyIdx = (LastEntropyIdx + 1) % NUM_ENTROPY_VALUES;
DecompressedBufferOffset += NumDecompressedSamples * sizeof(opus_int16);
CompressedBufferOffset += CompressedBufferSize;
LastCompressedOffset = CompressedOffsets[i];
}
}
else
{
UE_LOG(LogVoiceDecode, Warning, TEXT("Decompression buffer too small to decode voice"));
break;
}
}
OutRawDataSize = DecompressedBufferOffset;
}
else
{
UE_LOG(LogVoiceDecode, Warning, TEXT("Failed to decode: buffer corrupted"));
OutRawDataSize = 0;
}
UE_LOG(LogVoiceDecode, VeryVerbose, TEXT("OpusDecode[%d]: RawSize: %d CompressedSize: %d NumFramesEncoded: %d "), PacketGeneration, OutRawDataSize, CompressedDataSize, NumFramesToDecode);
LastGeneration = PacketGeneration;
}
static int opus_decode_frame(OpusDecoder *st, const unsigned char *data,
opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec)
{
void *silk_dec;
CELTDecoder *celt_dec;
int i, silk_ret=0, celt_ret=0;
ec_dec dec;
opus_int32 silk_frame_size;
int pcm_silk_size;
VARDECL(opus_int16, pcm_silk);
int pcm_transition_silk_size;
VARDECL(opus_val16, pcm_transition_silk);
int pcm_transition_celt_size;
VARDECL(opus_val16, pcm_transition_celt);
opus_val16 *pcm_transition=NULL;
int redundant_audio_size;
VARDECL(opus_val16, redundant_audio);
int audiosize;
int mode;
int transition=0;
int start_band;
int redundancy=0;
int redundancy_bytes = 0;
int celt_to_silk=0;
int c;
int F2_5, F5, F10, F20;
const opus_val16 *window;
opus_uint32 redundant_rng = 0;
int celt_accum;
ALLOC_STACK;
silk_dec = (char*)st+st->silk_dec_offset;
celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset);
F20 = st->Fs/50;
F10 = F20>>1;
F5 = F10>>1;
F2_5 = F5>>1;
if (frame_size < F2_5)
{
RESTORE_STACK;
return OPUS_BUFFER_TOO_SMALL;
}
/* Limit frame_size to avoid excessive stack allocations. */
frame_size = IMIN(frame_size, st->Fs/25*3);
/* Payloads of 1 (2 including ToC) or 0 trigger the PLC/DTX */
if (len<=1)
{
data = NULL;
/* In that case, don't conceal more than what the ToC says */
frame_size = IMIN(frame_size, st->frame_size);
}
if (data != NULL)
{
audiosize = st->frame_size;
mode = st->mode;
ec_dec_init(&dec,(unsigned char*)data,len);
} else {
audiosize = frame_size;
mode = st->prev_mode;
if (mode == 0)
{
/* If we haven't got any packet yet, all we can do is return zeros */
for (i=0; i<audiosize*st->channels; i++)
pcm[i] = 0;
RESTORE_STACK;
return audiosize;
}
/* Avoids trying to run the PLC on sizes other than 2.5 (CELT), 5 (CELT),
10, or 20 (e.g. 12.5 or 30 ms). */
if (audiosize > F20)
{
do {
int ret = opus_decode_frame(st, NULL, 0, pcm, IMIN(audiosize, F20), 0);
if (ret<0)
{
RESTORE_STACK;
return ret;
}
pcm += ret*st->channels;
audiosize -= ret;
} while (audiosize > 0);
RESTORE_STACK;
return frame_size;
} else if (audiosize < F20)
{
if (audiosize > F10)
audiosize = F10;
else if (mode != MODE_SILK_ONLY && audiosize > F5 && audiosize < F10)
audiosize = F5;
}
}
/* In fixed-point, we can tell CELT to do the accumulation on top of the
SILK PCM buffer. This saves some stack space. */
#ifdef FIXED_POINT
celt_accum = (mode != MODE_CELT_ONLY) && (frame_size >= F10);
#else
celt_accum = 0;
#endif
pcm_transition_silk_size = ALLOC_NONE;
pcm_transition_celt_size = ALLOC_NONE;
if (data!=NULL && st->prev_mode > 0 && (
(mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY && !st->prev_redundancy)
|| (mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) )
)
{
transition = 1;
/* Decide where to allocate the stack memory for pcm_transition */
if (mode == MODE_CELT_ONLY)
pcm_transition_celt_size = F5*st->channels;
else
pcm_transition_silk_size = F5*st->channels;
}
ALLOC(pcm_transition_celt, pcm_transition_celt_size, opus_val16);
if (transition && mode == MODE_CELT_ONLY)
{
pcm_transition = pcm_transition_celt;
opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0);
}
if (audiosize > frame_size)
{
/*fprintf(stderr, "PCM buffer too small: %d vs %d (mode = %d)\n", audiosize, frame_size, mode);*/
RESTORE_STACK;
return OPUS_BAD_ARG;
} else {
frame_size = audiosize;
}
/* Don't allocate any memory when in CELT-only mode */
pcm_silk_size = (mode != MODE_CELT_ONLY && !celt_accum) ? IMAX(F10, frame_size)*st->channels : ALLOC_NONE;
ALLOC(pcm_silk, pcm_silk_size, opus_int16);
/* SILK processing */
if (mode != MODE_CELT_ONLY)
{
int lost_flag, decoded_samples;
opus_int16 *pcm_ptr;
#ifdef FIXED_POINT
if (celt_accum)
pcm_ptr = pcm;
else
#endif
pcm_ptr = pcm_silk;
if (st->prev_mode==MODE_CELT_ONLY)
silk_InitDecoder( silk_dec );
/* The SILK PLC cannot produce frames of less than 10 ms */
st->DecControl.payloadSize_ms = IMAX(10, 1000 * audiosize / st->Fs);
if (data != NULL)
{
st->DecControl.nChannelsInternal = st->stream_channels;
if( mode == MODE_SILK_ONLY ) {
if( st->bandwidth == OPUS_BANDWIDTH_NARROWBAND ) {
st->DecControl.internalSampleRate = 8000;
} else if( st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND ) {
st->DecControl.internalSampleRate = 12000;
} else if( st->bandwidth == OPUS_BANDWIDTH_WIDEBAND ) {
st->DecControl.internalSampleRate = 16000;
} else {
st->DecControl.internalSampleRate = 16000;
silk_assert( 0 );
}
} else {
/* Hybrid mode */
st->DecControl.internalSampleRate = 16000;
}
}
lost_flag = data == NULL ? 1 : 2 * decode_fec;
decoded_samples = 0;
do {
/* Call SILK decoder */
int first_frame = decoded_samples == 0;
silk_ret = silk_Decode( silk_dec, &st->DecControl,
lost_flag, first_frame, &dec, pcm_ptr, &silk_frame_size );
if( silk_ret ) {
if (lost_flag) {
/* PLC failure should not be fatal */
silk_frame_size = frame_size;
for (i=0; i<frame_size*st->channels; i++)
pcm_ptr[i] = 0;
} else {
RESTORE_STACK;
return OPUS_INTERNAL_ERROR;
}
}
pcm_ptr += silk_frame_size * st->channels;
decoded_samples += silk_frame_size;
} while( decoded_samples < frame_size );
}
start_band = 0;
if (!decode_fec && mode != MODE_CELT_ONLY && data != NULL
&& ec_tell(&dec)+17+20*(st->mode == MODE_HYBRID) <= 8*len)
{
/* Check if we have a redundant 0-8 kHz band */
if (mode == MODE_HYBRID)
redundancy = ec_dec_bit_logp(&dec, 12);
else
redundancy = 1;
if (redundancy)
{
celt_to_silk = ec_dec_bit_logp(&dec, 1);
/* redundancy_bytes will be at least two, in the non-hybrid
case due to the ec_tell() check above */
redundancy_bytes = mode==MODE_HYBRID ?
(opus_int32)ec_dec_uint(&dec, 256)+2 :
len-((ec_tell(&dec)+7)>>3);
len -= redundancy_bytes;
/* This is a sanity check. It should never happen for a valid
packet, so the exact behaviour is not normative. */
if (len*8 < ec_tell(&dec))
{
len = 0;
redundancy_bytes = 0;
redundancy = 0;
}
/* Shrink decoder because of raw bits */
dec.storage -= redundancy_bytes;
}
}
if (mode != MODE_CELT_ONLY)
start_band = 17;
{
int endband=21;
switch(st->bandwidth)
{
case OPUS_BANDWIDTH_NARROWBAND:
endband = 13;
break;
case OPUS_BANDWIDTH_MEDIUMBAND:
case OPUS_BANDWIDTH_WIDEBAND:
endband = 17;
break;
case OPUS_BANDWIDTH_SUPERWIDEBAND:
endband = 19;
break;
case OPUS_BANDWIDTH_FULLBAND:
endband = 21;
break;
}
celt_decoder_ctl(celt_dec, CELT_SET_END_BAND(endband));
celt_decoder_ctl(celt_dec, CELT_SET_CHANNELS(st->stream_channels));
}
if (redundancy)
{
transition = 0;
pcm_transition_silk_size=ALLOC_NONE;
}
ALLOC(pcm_transition_silk, pcm_transition_silk_size, opus_val16);
if (transition && mode != MODE_CELT_ONLY)
{
pcm_transition = pcm_transition_silk;
opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0);
}
/* Only allocation memory for redundancy if/when needed */
redundant_audio_size = redundancy ? F5*st->channels : ALLOC_NONE;
ALLOC(redundant_audio, redundant_audio_size, opus_val16);
/* 5 ms redundant frame for CELT->SILK*/
if (redundancy && celt_to_silk)
{
celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0));
celt_decode_with_ec(celt_dec, data+len, redundancy_bytes,
redundant_audio, F5, NULL, 0);
celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng));
}
/* MUST be after PLC */
celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(start_band));
if (mode != MODE_SILK_ONLY)
{
int celt_frame_size = IMIN(F20, frame_size);
/* Make sure to discard any previous CELT state */
if (mode != st->prev_mode && st->prev_mode > 0 && !st->prev_redundancy)
celt_decoder_ctl(celt_dec, OPUS_RESET_STATE);
/* Decode CELT */
celt_ret = celt_decode_with_ec(celt_dec, decode_fec ? NULL : data,
len, pcm, celt_frame_size, &dec, celt_accum);
} else {
unsigned char silence[2] = {0xFF, 0xFF};
if (!celt_accum)
{
for (i=0; i<frame_size*st->channels; i++)
pcm[i] = 0;
}
/* For hybrid -> SILK transitions, we let the CELT MDCT
do a fade-out by decoding a silence frame */
if (st->prev_mode == MODE_HYBRID && !(redundancy && celt_to_silk && st->prev_redundancy) )
{
celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0));
celt_decode_with_ec(celt_dec, silence, 2, pcm, F2_5, NULL, celt_accum);
}
}
if (mode != MODE_CELT_ONLY && !celt_accum)
{
#ifdef FIXED_POINT
for (i=0; i<frame_size*st->channels; i++)
pcm[i] = SAT16(ADD32(pcm[i], pcm_silk[i]));
#else
for (i=0; i<frame_size*st->channels; i++)
pcm[i] = pcm[i] + (opus_val16)((1.f/32768.f)*pcm_silk[i]);
#endif
}
{
const CELTMode *celt_mode;
celt_decoder_ctl(celt_dec, CELT_GET_MODE(&celt_mode));
window = celt_mode->window;
}
/* 5 ms redundant frame for SILK->CELT */
if (redundancy && !celt_to_silk)
{
celt_decoder_ctl(celt_dec, OPUS_RESET_STATE);
celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0));
celt_decode_with_ec(celt_dec, data+len, redundancy_bytes, redundant_audio, F5, NULL, 0);
celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng));
smooth_fade(pcm+st->channels*(frame_size-F2_5), redundant_audio+st->channels*F2_5,
pcm+st->channels*(frame_size-F2_5), F2_5, st->channels, window, st->Fs);
}
if (redundancy && celt_to_silk)
{
for (c=0; c<st->channels; c++)
{
for (i=0; i<F2_5; i++)
pcm[st->channels*i+c] = redundant_audio[st->channels*i+c];
}
smooth_fade(redundant_audio+st->channels*F2_5, pcm+st->channels*F2_5,
pcm+st->channels*F2_5, F2_5, st->channels, window, st->Fs);
}
if (transition)
{
if (audiosize >= F5)
{
for (i=0; i<st->channels*F2_5; i++)
pcm[i] = pcm_transition[i];
smooth_fade(pcm_transition+st->channels*F2_5, pcm+st->channels*F2_5,
pcm+st->channels*F2_5, F2_5,
st->channels, window, st->Fs);
} else {
/* Not enough time to do a clean transition, but we do it anyway
This will not preserve amplitude perfectly and may introduce
a bit of temporal aliasing, but it shouldn't be too bad and
that's pretty much the best we can do. In any case, generating this
transition it pretty silly in the first place */
smooth_fade(pcm_transition, pcm,
pcm, F2_5,
st->channels, window, st->Fs);
}
}
if(st->decode_gain)
{
opus_val32 gain;
gain = celt_exp2(MULT16_16_P15(QCONST16(6.48814081e-4f, 25), st->decode_gain));
for (i=0; i<frame_size*st->channels; i++)
{
opus_val32 x;
x = MULT16_32_P16(pcm[i],gain);
pcm[i] = SATURATE(x, 32767);
}
}
if (len <= 1)
st->rangeFinal = 0;
else
st->rangeFinal = dec.rng ^ redundant_rng;
st->prev_mode = mode;
st->prev_redundancy = redundancy && !celt_to_silk;
if (celt_ret>=0)
{
if (OPUS_CHECK_ARRAY(pcm, audiosize*st->channels))
OPUS_PRINT_INT(audiosize);
}
RESTORE_STACK;
return celt_ret < 0 ? celt_ret : audiosize;
}
int32 FVoiceEncoderOpus::Encode(const uint8* RawPCMData, uint32 RawDataSize, uint8* OutCompressedData, uint32& OutCompressedDataSize)
{
check(Encoder);
SCOPE_CYCLE_COUNTER(STAT_Voice_Encoding);
int32 HeaderSize = 0;
int32 BytesPerFrame = FrameSize * NumChannels * sizeof(opus_int16);
int32 MaxFramesEncoded = MAX_OPUS_UNCOMPRESSED_BUFFER_SIZE / BytesPerFrame;
// total bytes / bytes per frame
int32 NumFramesToEncode = FMath::Min((int32)RawDataSize / BytesPerFrame, MaxFramesEncoded);
int32 DataRemainder = RawDataSize % BytesPerFrame;
int32 RawDataStride = BytesPerFrame;
// Store the number of frames to be encoded
check(NumFramesToEncode < MAX_uint8);
OutCompressedData[0] = NumFramesToEncode;
OutCompressedData[1] = Generation;
HeaderSize += 2 * sizeof(uint8);
// Store the offset to each encoded frame
uint16* CompressedOffsets = (uint16*)(OutCompressedData + HeaderSize);
uint32 LengthOfCompressedOffsets = NumFramesToEncode * sizeof(uint16);
HeaderSize += LengthOfCompressedOffsets;
// Store the entropy to each encoded frame
uint32 LengthOfEntropyOffsets = 0;
#if ADD_ENTROPY_TO_PACKET
uint32* EntropyOffsets = (uint32*)(OutCompressedData + HeaderSize);
LengthOfEntropyOffsets = NumFramesToEncode * sizeof(uint32);
#endif
HeaderSize += LengthOfEntropyOffsets;
// Space available after overhead
int32 AvailableBufferSize = OutCompressedDataSize - HeaderSize;
// Start of the actual compressed data
uint8* CompressedDataStart = OutCompressedData + HeaderSize;
int32 CompressedBufferOffset = 0;
for (int32 i = 0; i < NumFramesToEncode; i++)
{
int32 CompressedLength = 0;
CompressedLength = opus_encode(Encoder, (const opus_int16*)(RawPCMData + (i * RawDataStride)), FrameSize, CompressedDataStart + CompressedBufferOffset, AvailableBufferSize);
if (CompressedLength < 0)
{
const char* ErrorStr = opus_strerror(CompressedLength);
UE_LOG(LogVoiceEncode, Warning, TEXT("Failed to encode: [%d] %s"), CompressedLength, ANSI_TO_TCHAR(ErrorStr));
OutCompressedData[0] = 0;
OutCompressedDataSize = 0;
return 0;
}
else if (CompressedLength != 1)
{
opus_encoder_ctl(Encoder, OPUS_GET_FINAL_RANGE(&Entropy[LastEntropyIdx]));
#if ADD_ENTROPY_TO_PACKET
UE_LOG(LogVoiceEncode, VeryVerbose, TEXT("Entropy[%d]=%d"), i, Entropy[LastEntropyIdx]);
EntropyOffsets[i] = Entropy[LastEntropyIdx];
#endif
LastEntropyIdx = (LastEntropyIdx + 1) % NUM_ENTROPY_VALUES;
#if DEBUG_OPUS
DebugFrameEncodeInfo(CompressedDataStart + CompressedBufferOffset, CompressedLength, SampleRate);
#endif // DEBUG_OPUS
AvailableBufferSize -= CompressedLength;
CompressedBufferOffset += CompressedLength;
check(CompressedBufferOffset < MAX_uint16);
CompressedOffsets[i] = (uint16)CompressedBufferOffset;
}
else
{
UE_LOG(LogVoiceEncode, Warning, TEXT("Nothing to encode!"));
}
}
// End of buffer
OutCompressedDataSize = HeaderSize + CompressedBufferOffset;
UE_LOG(LogVoiceEncode, VeryVerbose, TEXT("OpusEncode[%d]: RawSize: %d CompressedSize: %d NumFramesEncoded: %d Remains: %d"), Generation, RawDataSize, OutCompressedDataSize, NumFramesToEncode, DataRemainder);
Generation = (Generation + 1) % MAX_uint8;
return DataRemainder;
}
int main(int argc, char **argv)
{
int c;
int option_index = 0;
char *inFile, *outFile;
FILE *fin, *fout=NULL, *frange=NULL;
float *output;
int frame_size=0;
OpusMSDecoder *st=NULL;
opus_int64 packet_count=0;
int total_links=0;
int stream_init = 0;
int quiet = 0;
ogg_int64_t page_granule=0;
ogg_int64_t link_out=0;
struct option long_options[] =
{
{"help", no_argument, NULL, 0},
{"quiet", no_argument, NULL, 0},
{"version", no_argument, NULL, 0},
{"version-short", no_argument, NULL, 0},
{"rate", required_argument, NULL, 0},
{"gain", required_argument, NULL, 0},
{"no-dither", no_argument, NULL, 0},
{"packet-loss", required_argument, NULL, 0},
{"save-range", required_argument, NULL, 0},
{0, 0, 0, 0}
};
ogg_sync_state oy;
ogg_page og;
ogg_packet op;
ogg_stream_state os;
int close_in=0;
int eos=0;
ogg_int64_t audio_size=0;
double last_coded_seconds=0;
float loss_percent=-1;
float manual_gain=0;
int channels=-1;
int mapping_family;
int rate=0;
int wav_format=0;
int preskip=0;
int gran_offset=0;
int has_opus_stream=0;
ogg_int32_t opus_serialno;
int dither=1;
shapestate shapemem;
SpeexResamplerState *resampler=NULL;
float gain=1;
int streams=0;
size_t last_spin=0;
#ifdef WIN_UNICODE
int argc_utf8;
char **argv_utf8;
#endif
if(query_cpu_support()){
fprintf(stderr,"\n\n** WARNING: This program with compiled with SSE%s\n",query_cpu_support()>1?"2":"");
fprintf(stderr," but this CPU claims to lack these instructions. **\n\n");
}
#ifdef WIN_UNICODE
(void)argc;
(void)argv;
init_console_utf8();
init_commandline_arguments_utf8(&argc_utf8, &argv_utf8);
#endif
output=0;
shapemem.a_buf=0;
shapemem.b_buf=0;
shapemem.mute=960;
shapemem.fs=0;
/*Process options*/
while(1)
{
c = getopt_long (argc_utf8, argv_utf8, "hV",
long_options, &option_index);
if (c==-1)
break;
switch(c)
{
case 0:
if (strcmp(long_options[option_index].name,"help")==0)
{
usage();
quit(0);
} else if (strcmp(long_options[option_index].name,"quiet")==0)
{
quiet = 1;
} else if (strcmp(long_options[option_index].name,"version")==0)
{
version();
quit(0);
} else if (strcmp(long_options[option_index].name,"version-short")==0)
{
version_short();
quit(0);
} else if (strcmp(long_options[option_index].name,"no-dither")==0)
{
dither=0;
} else if (strcmp(long_options[option_index].name,"rate")==0)
{
rate=atoi (optarg);
} else if (strcmp(long_options[option_index].name,"gain")==0)
{
manual_gain=atof (optarg);
}else if(strcmp(long_options[option_index].name,"save-range")==0){
frange=fopen_utf8(optarg,"w");
if(frange==NULL){
perror(optarg);
fprintf(stderr,"Could not open save-range file: %s\n",optarg);
fprintf(stderr,"Must provide a writable file name.\n");
quit(1);
}
} else if (strcmp(long_options[option_index].name,"packet-loss")==0)
{
loss_percent = atof(optarg);
}
break;
case 'h':
usage();
quit(0);
break;
case 'V':
version();
quit(0);
break;
case '?':
usage();
quit(1);
break;
}
}
if (argc_utf8-optind!=2 && argc_utf8-optind!=1)
{
usage();
quit(1);
}
inFile=argv_utf8[optind];
/*Output to a file or playback?*/
if (argc_utf8-optind==2){
/*If we're outputting to a file, should we apply a wav header?*/
int i;
char *ext;
outFile=argv_utf8[optind+1];
ext=".wav";
i=strlen(outFile)-4;
wav_format=i>=0;
while(wav_format&&ext&&outFile[i]) {
wav_format&=tolower(outFile[i++])==*ext++;
}
}else {
outFile="";
wav_format=0;
/*If playing to audio out, default the rate to 48000
instead of the original rate. The original rate is
only important for minimizing surprise about the rate
of output files and preserving length, which aren't
relevant for playback. Many audio devices sound
better at 48kHz and not resampling also saves CPU.*/
if(rate==0)rate=48000;
}
/*Open input file*/
if (strcmp(inFile, "-")==0)
{
#if defined WIN32 || defined _WIN32
_setmode(_fileno(stdin), _O_BINARY);
#endif
fin=stdin;
}
else
{
fin = fopen_utf8(inFile, "rb");
if (!fin)
{
perror(inFile);
quit(1);
}
close_in=1;
}
/* .opus files use the Ogg container to provide framing and timekeeping.
* http://tools.ietf.org/html/draft-terriberry-oggopus
* The easiest way to decode the Ogg container is to use libogg, so
* thats what we do here.
* Using libogg is fairly straight forward-- you take your stream of bytes
* and feed them to ogg_sync_ and it periodically returns Ogg pages, you
* check if the pages belong to the stream you're decoding then you give
* them to libogg and it gives you packets. You decode the packets. The
* pages also provide timing information.*/
ogg_sync_init(&oy);
/*Main decoding loop*/
while (1)
{
char *data;
int i, nb_read;
/*Get the ogg buffer for writing*/
data = ogg_sync_buffer(&oy, 200);
/*Read bitstream from input file*/
nb_read = fread(data, sizeof(char), 200, fin);
ogg_sync_wrote(&oy, nb_read);
/*Loop for all complete pages we got (most likely only one)*/
while (ogg_sync_pageout(&oy, &og)==1)
{
if (stream_init == 0) {
ogg_stream_init(&os, ogg_page_serialno(&og));
stream_init = 1;
}
if (ogg_page_serialno(&og) != os.serialno) {
/* so all streams are read. */
ogg_stream_reset_serialno(&os, ogg_page_serialno(&og));
}
/*Add page to the bitstream*/
ogg_stream_pagein(&os, &og);
page_granule = ogg_page_granulepos(&og);
/*Extract all available packets*/
while (ogg_stream_packetout(&os, &op) == 1)
{
/*OggOpus streams are identified by a magic string in the initial
stream header.*/
if (op.b_o_s && op.bytes>=8 && !memcmp(op.packet, "OpusHead", 8)) {
if(!has_opus_stream)
{
opus_serialno = os.serialno;
has_opus_stream = 1;
link_out = 0;
packet_count = 0;
eos = 0;
total_links++;
} else {
fprintf(stderr,"Warning: ignoring opus stream %" I64FORMAT "\n",(long long)os.serialno);
}
}
if (!has_opus_stream || os.serialno != opus_serialno)
break;
/*If first packet in a logical stream, process the Opus header*/
if (packet_count==0)
{
st = process_header(&op, &rate, &mapping_family, &channels, &preskip, &gain, manual_gain, &streams, wav_format, quiet);
if (!st)
quit(1);
/*Remember how many samples at the front we were told to skip
so that we can adjust the timestamp counting.*/
gran_offset=preskip;
/*Setup the memory for the dithered output*/
if(!shapemem.a_buf)
{
shapemem.a_buf=calloc(channels,sizeof(float)*4);
shapemem.b_buf=calloc(channels,sizeof(float)*4);
shapemem.fs=rate;
}
if(!output)output=malloc(sizeof(float)*MAX_FRAME_SIZE*channels);
/*Normal players should just play at 48000 or their maximum rate,
as described in the OggOpus spec. But for commandline tools
like opusdec it can be desirable to exactly preserve the original
sampling rate and duration, so we have a resampler here.*/
if (rate != 48000)
{
int err;
resampler = speex_resampler_init(channels, 48000, rate, 5, &err);
if (err!=0)
fprintf(stderr, "resampler error: %s\n", speex_resampler_strerror(err));
speex_resampler_skip_zeros(resampler);
}
if(!fout)fout=out_file_open(outFile, &wav_format, rate, mapping_family, &channels);
} else if (packet_count==1)
{
if (!quiet)
print_comments((char*)op.packet, op.bytes);
} else {
int ret;
opus_int64 maxout;
opus_int64 outsamp;
int lost=0;
if (loss_percent>0 && 100*((float)rand())/RAND_MAX<loss_percent)
lost=1;
/*End of stream condition*/
if (op.e_o_s && os.serialno == opus_serialno)eos=1; /* don't care for anything except opus eos */
/*Are we simulating loss for this packet?*/
if (!lost){
/*Decode Opus packet*/
ret = opus_multistream_decode_float(st, (unsigned char*)op.packet, op.bytes, output, MAX_FRAME_SIZE, 0);
} else {
/*Extract the original duration.
Normally you wouldn't have it for a lost packet, but normally the
transports used on lossy channels will effectively tell you.
This avoids opusdec squaking when the decoded samples and
granpos mismatches.*/
opus_int32 lost_size;
lost_size = MAX_FRAME_SIZE;
if(op.bytes>0){
opus_int32 spp;
spp=opus_packet_get_nb_frames(op.packet,op.bytes);
if(spp>0){
spp*=opus_packet_get_samples_per_frame(op.packet,48000/*decoding_rate*/);
if(spp>0)lost_size=spp;
}
}
/*Invoke packet loss concealment.*/
ret = opus_multistream_decode_float(st, NULL, 0, output, lost_size, 0);
}
if(!quiet){
/*Display a progress spinner while decoding.*/
static const char spinner[]="|/-\\";
double coded_seconds = (double)audio_size/(channels*rate*sizeof(short));
if(coded_seconds>=last_coded_seconds+1){
fprintf(stderr,"\r[%c] %02d:%02d:%02d", spinner[last_spin&3],
(int)(coded_seconds/3600),(int)(coded_seconds/60)%60,
(int)(coded_seconds)%60);
fflush(stderr);
last_spin++;
last_coded_seconds=coded_seconds;
}
}
/*If the decoder returned less than zero, we have an error.*/
if (ret<0)
{
fprintf (stderr, "Decoding error: %s\n", opus_strerror(ret));
break;
}
frame_size = ret;
/*If we're collecting --save-range debugging data, collect it now.*/
if(frange!=NULL){
OpusDecoder *od;
opus_uint32 rngs[256];
for(i=0;i<streams;i++){
ret=opus_multistream_decoder_ctl(st,OPUS_MULTISTREAM_GET_DECODER_STATE(i,&od));
ret=opus_decoder_ctl(od,OPUS_GET_FINAL_RANGE(&rngs[i]));
}
save_range(frange,frame_size*(48000/48000/*decoding_rate*/),op.packet,op.bytes,
rngs,streams);
}
/*Apply header gain, if we're not using an opus library new
enough to do this internally.*/
if (gain!=0){
for (i=0;i<frame_size*channels;i++)
output[i] *= gain;
}
/*This handles making sure that our output duration respects
the final end-trim by not letting the output sample count
get ahead of the granpos indicated value.*/
maxout=((page_granule-gran_offset)*rate/48000)-link_out;
outsamp=audio_write(output, channels, frame_size, fout, resampler, &preskip, dither?&shapemem:0, strlen(outFile)!=0,0>maxout?0:maxout);
link_out+=outsamp;
audio_size+=sizeof(short)*outsamp*channels;
}
packet_count++;
}
/*We're done, drain the resampler if we were using it.*/
if(eos && resampler)
{
float *zeros;
int drain;
zeros=(float *)calloc(100*channels,sizeof(float));
drain = speex_resampler_get_input_latency(resampler);
do {
opus_int64 outsamp;
int tmp = drain;
if (tmp > 100)
tmp = 100;
outsamp=audio_write(zeros, channels, tmp, fout, resampler, NULL, &shapemem, strlen(outFile)!=0, ((page_granule-gran_offset)*rate/48000)-link_out);
link_out+=outsamp;
audio_size+=sizeof(short)*outsamp*channels;
drain -= tmp;
} while (drain>0);
free(zeros);
speex_resampler_destroy(resampler);
resampler=NULL;
}
if(eos)
{
has_opus_stream=0;
if(st)opus_multistream_decoder_destroy(st);
st=NULL;
}
}
if (feof(fin)) {
if(!quiet) {
fprintf(stderr, "\rDecoding complete. \n");
fflush(stderr);
}
break;
}
}
/*If we were writing wav, go set the duration.*/
if (strlen(outFile)!=0 && fout && wav_format>=0 && audio_size<0x7FFFFFFF)
{
if (fseek(fout,4,SEEK_SET)==0)
{
int tmp;
tmp = le_int(audio_size+20+wav_format);
if(fwrite(&tmp,4,1,fout)!=1)fprintf(stderr,"Error writing end length.\n");
if (fseek(fout,16+wav_format,SEEK_CUR)==0)
{
tmp = le_int(audio_size);
if(fwrite(&tmp,4,1,fout)!=1)fprintf(stderr,"Error writing header length.\n");
} else
{
fprintf (stderr, "First seek worked, second didn't\n");
}
} else {
fprintf (stderr, "Cannot seek on wave file, size will be incorrect\n");
}
}
/*Did we make it to the end without recovering ANY opus logical streams?*/
if(!total_links)fprintf (stderr, "This doesn't look like a Opus file\n");
if (stream_init)
ogg_stream_clear(&os);
ogg_sync_clear(&oy);
#if defined WIN32 || defined _WIN32
if (strlen(outFile)==0)
WIN_Audio_close ();
#endif
if(shapemem.a_buf)free(shapemem.a_buf);
if(shapemem.b_buf)free(shapemem.b_buf);
if(output)free(output);
if(frange)fclose(frange);
if (close_in)
fclose(fin);
if (fout != NULL)
fclose(fout);
#ifdef WIN_UNICODE
free_commandline_arguments_utf8(&argc_utf8, &argv_utf8);
uninit_console_utf8();
#endif
return 0;
}