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Diffstat (limited to 'media/libopus/src/opus_decoder.c')
| -rw-r--r-- | media/libopus/src/opus_decoder.c | 1041 |
1 files changed, 1041 insertions, 0 deletions
diff --git a/media/libopus/src/opus_decoder.c b/media/libopus/src/opus_decoder.c new file mode 100644 index 0000000000..6520e748ea --- /dev/null +++ b/media/libopus/src/opus_decoder.c @@ -0,0 +1,1041 @@ +/* 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; +}; + +#if defined(ENABLE_HARDENING) || defined(ENABLE_ASSERTIONS) +static void validate_opus_decoder(OpusDecoder *st) +{ + celt_assert(st->channels == 1 || st->channels == 2); + celt_assert(st->Fs == 48000 || st->Fs == 24000 || st->Fs == 16000 || st->Fs == 12000 || st->Fs == 8000); + celt_assert(st->DecControl.API_sampleRate == st->Fs); + celt_assert(st->DecControl.internalSampleRate == 0 || st->DecControl.internalSampleRate == 16000 || st->DecControl.internalSampleRate == 12000 || st->DecControl.internalSampleRate == 8000); + celt_assert(st->DecControl.nChannelsAPI == st->channels); + celt_assert(st->DecControl.nChannelsInternal == 0 || st->DecControl.nChannelsInternal == 1 || st->DecControl.nChannelsInternal == 2); + celt_assert(st->DecControl.payloadSize_ms == 0 || st->DecControl.payloadSize_ms == 10 || st->DecControl.payloadSize_ms == 20 || st->DecControl.payloadSize_ms == 40 || st->DecControl.payloadSize_ms == 60); +#ifdef OPUS_ARCHMASK + celt_assert(st->arch >= 0); + celt_assert(st->arch <= OPUS_ARCHMASK); +#endif + celt_assert(st->stream_channels == 1 || st->stream_channels == 2); +} +#define VALIDATE_OPUS_DECODER(st) validate_opus_decoder(st) +#else +#define VALIDATE_OPUS_DECODER(st) +#endif + +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 decoder */ + 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 bandwidth; + 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; + bandwidth = st->bandwidth; + ec_dec_init(&dec,(unsigned char*)data,len); + } else { + audiosize = frame_size; + /* Run PLC using last used mode (CELT if we ended with CELT redundancy) */ + mode = st->prev_redundancy ? MODE_CELT_ONLY : st->prev_mode; + bandwidth = 0; + + 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( bandwidth == OPUS_BANDWIDTH_NARROWBAND ) { + st->DecControl.internalSampleRate = 8000; + } else if( bandwidth == OPUS_BANDWIDTH_MEDIUMBAND ) { + st->DecControl.internalSampleRate = 12000; + } else if( bandwidth == OPUS_BANDWIDTH_WIDEBAND ) { + st->DecControl.internalSampleRate = 16000; + } else { + st->DecControl.internalSampleRate = 16000; + celt_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*(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; + + 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); + } + + + if (bandwidth) + { + int endband=21; + + switch(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; + default: + celt_assert(0); + break; + } + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_END_BAND(endband))); + } + MUST_SUCCEED(celt_decoder_ctl(celt_dec, CELT_SET_CHANNELS(st->stream_channels))); + + /* 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) + { + /* If the previous frame did not use CELT (the first redundancy frame in + a transition from SILK may have been lost) then the CELT decoder is + stale at this point and the redundancy audio is not useful, however + the final range is still needed (for testing), so the redundancy is + always decoded but the decoded audio may not be used */ + MUST_SUCCEED(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); + MUST_SUCCEED(celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng))); + } + + /* MUST be after PLC */ + MUST_SUCCEED(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) + MUST_SUCCEED(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) ) + { + MUST_SUCCEED(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; + MUST_SUCCEED(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) + { + MUST_SUCCEED(celt_decoder_ctl(celt_dec, OPUS_RESET_STATE)); + MUST_SUCCEED(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); + MUST_SUCCEED(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); + } + /* 5ms redundant frame for CELT->SILK; ignore if the previous frame did not + use CELT (the first redundancy frame in a transition from SILK may have + been lost) */ + if (redundancy && celt_to_silk && (st->prev_mode != MODE_SILK_ONLY || st->prev_redundancy)) + { + 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]; + VALIDATE_OPUS_DECODER(st); + 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; + } + celt_assert(st->channels == 1 || st->channels == 2); + 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; + } + celt_assert(st->channels == 1 || st->channels == 2); + 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) + ret = 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_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + *value = st->last_packet_duration; + } + break; + case OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + { + goto bad_arg; + } + ret = celt_decoder_ctl(celt_dec, OPUS_SET_PHASE_INVERSION_DISABLED(value)); + } + break; + case OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (!value) + { + goto bad_arg; + } + ret = celt_decoder_ctl(celt_dec, OPUS_GET_PHASE_INVERSION_DISABLED(value)); + } + 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); +} |