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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/libopus/src/opus_decoder.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/libopus/src/opus_decoder.c')
-rw-r--r--media/libopus/src/opus_decoder.c981
1 files changed, 981 insertions, 0 deletions
diff --git a/media/libopus/src/opus_decoder.c b/media/libopus/src/opus_decoder.c
new file mode 100644
index 000000000..080bec507
--- /dev/null
+++ b/media/libopus/src/opus_decoder.c
@@ -0,0 +1,981 @@
+/* Copyright (c) 2010 Xiph.Org Foundation, Skype Limited
+ Written by Jean-Marc Valin and Koen Vos */
+/*
+ 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.
+*/
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#ifndef OPUS_BUILD
+# error "OPUS_BUILD _MUST_ be defined to build Opus. This probably means you need other defines as well, as in a config.h. See the included build files for details."
+#endif
+
+#if defined(__GNUC__) && (__GNUC__ >= 2) && !defined(__OPTIMIZE__) && !defined(OPUS_WILL_BE_SLOW)
+# pragma message "You appear to be compiling without optimization, if so opus will be very slow."
+#endif
+
+#include <stdarg.h>
+#include "celt.h"
+#include "opus.h"
+#include "entdec.h"
+#include "modes.h"
+#include "API.h"
+#include "stack_alloc.h"
+#include "float_cast.h"
+#include "opus_private.h"
+#include "os_support.h"
+#include "structs.h"
+#include "define.h"
+#include "mathops.h"
+#include "cpu_support.h"
+
+struct OpusDecoder {
+ int celt_dec_offset;
+ int silk_dec_offset;
+ int channels;
+ opus_int32 Fs; /** Sampling rate (at the API level) */
+ silk_DecControlStruct DecControl;
+ int decode_gain;
+ int arch;
+
+ /* Everything beyond this point gets cleared on a reset */
+#define OPUS_DECODER_RESET_START stream_channels
+ int stream_channels;
+
+ int bandwidth;
+ int mode;
+ int prev_mode;
+ int frame_size;
+ int prev_redundancy;
+ int last_packet_duration;
+#ifndef FIXED_POINT
+ opus_val16 softclip_mem[2];
+#endif
+
+ opus_uint32 rangeFinal;
+};
+
+
+int opus_decoder_get_size(int channels)
+{
+ int silkDecSizeBytes, celtDecSizeBytes;
+ int ret;
+ if (channels<1 || channels > 2)
+ return 0;
+ ret = silk_Get_Decoder_Size( &silkDecSizeBytes );
+ if(ret)
+ return 0;
+ silkDecSizeBytes = align(silkDecSizeBytes);
+ celtDecSizeBytes = celt_decoder_get_size(channels);
+ return align(sizeof(OpusDecoder))+silkDecSizeBytes+celtDecSizeBytes;
+}
+
+int opus_decoder_init(OpusDecoder *st, opus_int32 Fs, int channels)
+{
+ void *silk_dec;
+ CELTDecoder *celt_dec;
+ int ret, silkDecSizeBytes;
+
+ if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)
+ || (channels!=1&&channels!=2))
+ return OPUS_BAD_ARG;
+
+ OPUS_CLEAR((char*)st, opus_decoder_get_size(channels));
+ /* Initialize SILK encoder */
+ ret = silk_Get_Decoder_Size(&silkDecSizeBytes);
+ if (ret)
+ return OPUS_INTERNAL_ERROR;
+
+ silkDecSizeBytes = align(silkDecSizeBytes);
+ st->silk_dec_offset = align(sizeof(OpusDecoder));
+ st->celt_dec_offset = st->silk_dec_offset+silkDecSizeBytes;
+ silk_dec = (char*)st+st->silk_dec_offset;
+ celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset);
+ st->stream_channels = st->channels = channels;
+
+ st->Fs = Fs;
+ st->DecControl.API_sampleRate = st->Fs;
+ st->DecControl.nChannelsAPI = st->channels;
+
+ /* Reset decoder */
+ ret = silk_InitDecoder( silk_dec );
+ if(ret)return OPUS_INTERNAL_ERROR;
+
+ /* Initialize CELT decoder */
+ ret = celt_decoder_init(celt_dec, Fs, channels);
+ if(ret!=OPUS_OK)return OPUS_INTERNAL_ERROR;
+
+ celt_decoder_ctl(celt_dec, CELT_SET_SIGNALLING(0));
+
+ st->prev_mode = 0;
+ st->frame_size = Fs/400;
+ st->arch = opus_select_arch();
+ return OPUS_OK;
+}
+
+OpusDecoder *opus_decoder_create(opus_int32 Fs, int channels, int *error)
+{
+ int ret;
+ OpusDecoder *st;
+ if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)
+ || (channels!=1&&channels!=2))
+ {
+ if (error)
+ *error = OPUS_BAD_ARG;
+ return NULL;
+ }
+ st = (OpusDecoder *)opus_alloc(opus_decoder_get_size(channels));
+ if (st == NULL)
+ {
+ if (error)
+ *error = OPUS_ALLOC_FAIL;
+ return NULL;
+ }
+ ret = opus_decoder_init(st, Fs, channels);
+ if (error)
+ *error = ret;
+ if (ret != OPUS_OK)
+ {
+ opus_free(st);
+ st = NULL;
+ }
+ return st;
+}
+
+static void smooth_fade(const opus_val16 *in1, const opus_val16 *in2,
+ opus_val16 *out, int overlap, int channels,
+ const opus_val16 *window, opus_int32 Fs)
+{
+ int i, c;
+ int inc = 48000/Fs;
+ for (c=0;c<channels;c++)
+ {
+ for (i=0;i<overlap;i++)
+ {
+ opus_val16 w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+ out[i*channels+c] = SHR32(MAC16_16(MULT16_16(w,in2[i*channels+c]),
+ Q15ONE-w, in1[i*channels+c]), 15);
+ }
+ }
+}
+
+static int opus_packet_get_mode(const unsigned char *data)
+{
+ int mode;
+ if (data[0]&0x80)
+ {
+ mode = MODE_CELT_ONLY;
+ } else if ((data[0]&0x60) == 0x60)
+ {
+ mode = MODE_HYBRID;
+ } else {
+ mode = MODE_SILK_ONLY;
+ }
+ return mode;
+}
+
+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, st->arch );
+ 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;
+
+}
+
+int opus_decode_native(OpusDecoder *st, const unsigned char *data,
+ opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec,
+ int self_delimited, opus_int32 *packet_offset, int soft_clip)
+{
+ int i, nb_samples;
+ int count, offset;
+ unsigned char toc;
+ int packet_frame_size, packet_bandwidth, packet_mode, packet_stream_channels;
+ /* 48 x 2.5 ms = 120 ms */
+ opus_int16 size[48];
+ if (decode_fec<0 || decode_fec>1)
+ return OPUS_BAD_ARG;
+ /* For FEC/PLC, frame_size has to be to have a multiple of 2.5 ms */
+ if ((decode_fec || len==0 || data==NULL) && frame_size%(st->Fs/400)!=0)
+ return OPUS_BAD_ARG;
+ if (len==0 || data==NULL)
+ {
+ int pcm_count=0;
+ do {
+ int ret;
+ ret = opus_decode_frame(st, NULL, 0, pcm+pcm_count*st->channels, frame_size-pcm_count, 0);
+ if (ret<0)
+ return ret;
+ pcm_count += ret;
+ } while (pcm_count < frame_size);
+ celt_assert(pcm_count == frame_size);
+ if (OPUS_CHECK_ARRAY(pcm, pcm_count*st->channels))
+ OPUS_PRINT_INT(pcm_count);
+ st->last_packet_duration = pcm_count;
+ return pcm_count;
+ } else if (len<0)
+ return OPUS_BAD_ARG;
+
+ packet_mode = opus_packet_get_mode(data);
+ packet_bandwidth = opus_packet_get_bandwidth(data);
+ packet_frame_size = opus_packet_get_samples_per_frame(data, st->Fs);
+ packet_stream_channels = opus_packet_get_nb_channels(data);
+
+ count = opus_packet_parse_impl(data, len, self_delimited, &toc, NULL,
+ size, &offset, packet_offset);
+ if (count<0)
+ return count;
+
+ data += offset;
+
+ if (decode_fec)
+ {
+ int duration_copy;
+ int ret;
+ /* If no FEC can be present, run the PLC (recursive call) */
+ if (frame_size < packet_frame_size || packet_mode == MODE_CELT_ONLY || st->mode == MODE_CELT_ONLY)
+ return opus_decode_native(st, NULL, 0, pcm, frame_size, 0, 0, NULL, soft_clip);
+ /* Otherwise, run the PLC on everything except the size for which we might have FEC */
+ duration_copy = st->last_packet_duration;
+ if (frame_size-packet_frame_size!=0)
+ {
+ ret = opus_decode_native(st, NULL, 0, pcm, frame_size-packet_frame_size, 0, 0, NULL, soft_clip);
+ if (ret<0)
+ {
+ st->last_packet_duration = duration_copy;
+ return ret;
+ }
+ celt_assert(ret==frame_size-packet_frame_size);
+ }
+ /* Complete with FEC */
+ st->mode = packet_mode;
+ st->bandwidth = packet_bandwidth;
+ st->frame_size = packet_frame_size;
+ st->stream_channels = packet_stream_channels;
+ ret = opus_decode_frame(st, data, size[0], pcm+st->channels*(frame_size-packet_frame_size),
+ packet_frame_size, 1);
+ if (ret<0)
+ return ret;
+ else {
+ if (OPUS_CHECK_ARRAY(pcm, frame_size*st->channels))
+ OPUS_PRINT_INT(frame_size);
+ st->last_packet_duration = frame_size;
+ return frame_size;
+ }
+ }
+
+ if (count*packet_frame_size > frame_size)
+ return OPUS_BUFFER_TOO_SMALL;
+
+ /* Update the state as the last step to avoid updating it on an invalid packet */
+ st->mode = packet_mode;
+ st->bandwidth = packet_bandwidth;
+ st->frame_size = packet_frame_size;
+ st->stream_channels = packet_stream_channels;
+
+ nb_samples=0;
+ for (i=0;i<count;i++)
+ {
+ int ret;
+ ret = opus_decode_frame(st, data, size[i], pcm+nb_samples*st->channels, frame_size-nb_samples, 0);
+ if (ret<0)
+ return ret;
+ celt_assert(ret==packet_frame_size);
+ data += size[i];
+ nb_samples += ret;
+ }
+ st->last_packet_duration = nb_samples;
+ if (OPUS_CHECK_ARRAY(pcm, nb_samples*st->channels))
+ OPUS_PRINT_INT(nb_samples);
+#ifndef FIXED_POINT
+ if (soft_clip)
+ opus_pcm_soft_clip(pcm, nb_samples, st->channels, st->softclip_mem);
+ else
+ st->softclip_mem[0]=st->softclip_mem[1]=0;
+#endif
+ return nb_samples;
+}
+
+#ifdef FIXED_POINT
+
+int opus_decode(OpusDecoder *st, const unsigned char *data,
+ opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec)
+{
+ if(frame_size<=0)
+ return OPUS_BAD_ARG;
+ return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL, 0);
+}
+
+#ifndef DISABLE_FLOAT_API
+int opus_decode_float(OpusDecoder *st, const unsigned char *data,
+ opus_int32 len, float *pcm, int frame_size, int decode_fec)
+{
+ VARDECL(opus_int16, out);
+ int ret, i;
+ int nb_samples;
+ ALLOC_STACK;
+
+ if(frame_size<=0)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+ if (data != NULL && len > 0 && !decode_fec)
+ {
+ nb_samples = opus_decoder_get_nb_samples(st, data, len);
+ if (nb_samples>0)
+ frame_size = IMIN(frame_size, nb_samples);
+ else
+ return OPUS_INVALID_PACKET;
+ }
+ ALLOC(out, frame_size*st->channels, opus_int16);
+
+ ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 0);
+ if (ret > 0)
+ {
+ for (i=0;i<ret*st->channels;i++)
+ pcm[i] = (1.f/32768.f)*(out[i]);
+ }
+ RESTORE_STACK;
+ return ret;
+}
+#endif
+
+
+#else
+int opus_decode(OpusDecoder *st, const unsigned char *data,
+ opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec)
+{
+ VARDECL(float, out);
+ int ret, i;
+ int nb_samples;
+ ALLOC_STACK;
+
+ if(frame_size<=0)
+ {
+ RESTORE_STACK;
+ return OPUS_BAD_ARG;
+ }
+
+ if (data != NULL && len > 0 && !decode_fec)
+ {
+ nb_samples = opus_decoder_get_nb_samples(st, data, len);
+ if (nb_samples>0)
+ frame_size = IMIN(frame_size, nb_samples);
+ else
+ return OPUS_INVALID_PACKET;
+ }
+ ALLOC(out, frame_size*st->channels, float);
+
+ ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 1);
+ if (ret > 0)
+ {
+ for (i=0;i<ret*st->channels;i++)
+ pcm[i] = FLOAT2INT16(out[i]);
+ }
+ RESTORE_STACK;
+ return ret;
+}
+
+int opus_decode_float(OpusDecoder *st, const unsigned char *data,
+ opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec)
+{
+ if(frame_size<=0)
+ return OPUS_BAD_ARG;
+ return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL, 0);
+}
+
+#endif
+
+int opus_decoder_ctl(OpusDecoder *st, int request, ...)
+{
+ int ret = OPUS_OK;
+ va_list ap;
+ void *silk_dec;
+ CELTDecoder *celt_dec;
+
+ silk_dec = (char*)st+st->silk_dec_offset;
+ celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset);
+
+
+ va_start(ap, request);
+
+ switch (request)
+ {
+ case OPUS_GET_BANDWIDTH_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->bandwidth;
+ }
+ break;
+ case OPUS_GET_FINAL_RANGE_REQUEST:
+ {
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->rangeFinal;
+ }
+ break;
+ case OPUS_RESET_STATE:
+ {
+ OPUS_CLEAR((char*)&st->OPUS_DECODER_RESET_START,
+ sizeof(OpusDecoder)-
+ ((char*)&st->OPUS_DECODER_RESET_START - (char*)st));
+
+ celt_decoder_ctl(celt_dec, OPUS_RESET_STATE);
+ silk_InitDecoder( silk_dec );
+ st->stream_channels = st->channels;
+ st->frame_size = st->Fs/400;
+ }
+ break;
+ case OPUS_GET_SAMPLE_RATE_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->Fs;
+ }
+ break;
+ case OPUS_GET_PITCH_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ if (st->prev_mode == MODE_CELT_ONLY)
+ celt_decoder_ctl(celt_dec, OPUS_GET_PITCH(value));
+ else
+ *value = st->DecControl.prevPitchLag;
+ }
+ break;
+ case OPUS_GET_GAIN_REQUEST:
+ {
+ opus_int32 *value = va_arg(ap, opus_int32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->decode_gain;
+ }
+ break;
+ case OPUS_SET_GAIN_REQUEST:
+ {
+ opus_int32 value = va_arg(ap, opus_int32);
+ if (value<-32768 || value>32767)
+ {
+ goto bad_arg;
+ }
+ st->decode_gain = value;
+ }
+ break;
+ case OPUS_GET_LAST_PACKET_DURATION_REQUEST:
+ {
+ opus_uint32 *value = va_arg(ap, opus_uint32*);
+ if (!value)
+ {
+ goto bad_arg;
+ }
+ *value = st->last_packet_duration;
+ }
+ break;
+ default:
+ /*fprintf(stderr, "unknown opus_decoder_ctl() request: %d", request);*/
+ ret = OPUS_UNIMPLEMENTED;
+ break;
+ }
+
+ va_end(ap);
+ return ret;
+bad_arg:
+ va_end(ap);
+ return OPUS_BAD_ARG;
+}
+
+void opus_decoder_destroy(OpusDecoder *st)
+{
+ opus_free(st);
+}
+
+
+int opus_packet_get_bandwidth(const unsigned char *data)
+{
+ int bandwidth;
+ if (data[0]&0x80)
+ {
+ bandwidth = OPUS_BANDWIDTH_MEDIUMBAND + ((data[0]>>5)&0x3);
+ if (bandwidth == OPUS_BANDWIDTH_MEDIUMBAND)
+ bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+ } else if ((data[0]&0x60) == 0x60)
+ {
+ bandwidth = (data[0]&0x10) ? OPUS_BANDWIDTH_FULLBAND :
+ OPUS_BANDWIDTH_SUPERWIDEBAND;
+ } else {
+ bandwidth = OPUS_BANDWIDTH_NARROWBAND + ((data[0]>>5)&0x3);
+ }
+ return bandwidth;
+}
+
+int opus_packet_get_nb_channels(const unsigned char *data)
+{
+ return (data[0]&0x4) ? 2 : 1;
+}
+
+int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len)
+{
+ int count;
+ if (len<1)
+ return OPUS_BAD_ARG;
+ count = packet[0]&0x3;
+ if (count==0)
+ return 1;
+ else if (count!=3)
+ return 2;
+ else if (len<2)
+ return OPUS_INVALID_PACKET;
+ else
+ return packet[1]&0x3F;
+}
+
+int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len,
+ opus_int32 Fs)
+{
+ int samples;
+ int count = opus_packet_get_nb_frames(packet, len);
+
+ if (count<0)
+ return count;
+
+ samples = count*opus_packet_get_samples_per_frame(packet, Fs);
+ /* Can't have more than 120 ms */
+ if (samples*25 > Fs*3)
+ return OPUS_INVALID_PACKET;
+ else
+ return samples;
+}
+
+int opus_decoder_get_nb_samples(const OpusDecoder *dec,
+ const unsigned char packet[], opus_int32 len)
+{
+ return opus_packet_get_nb_samples(packet, len, dec->Fs);
+}