/* declpcm.c Copyright (c) 2003-2019 HandBrake Team This file is part of the HandBrake source code Homepage: . It may be used under the terms of the GNU General Public License v2. For full terms see the file COPYING file or visit http://www.gnu.org/licenses/gpl-2.0.html */ #include "hb.h" #include "hbffmpeg.h" #include "audio_resample.h" struct hb_work_private_s { hb_job_t *job; uint32_t size; /* frame size in bytes */ uint32_t nchunks; /* number of samples pairs if paired */ uint32_t nsamples; /* frame size in samples */ uint32_t pos; /* buffer offset for next input data */ int64_t next_pts; /* pts for next output frame */ int scr_sequence; /* the following is frame info for the frame we're currently accumulating */ uint64_t duration; /* frame duration (in 90KHz ticks) */ uint32_t offset; /* where in buf frame starts */ uint32_t samplerate; /* sample rate in bits/sec */ uint8_t nchannels; uint8_t sample_size; /* bits per sample */ uint8_t frame[HB_DVD_READ_BUFFER_SIZE*2]; uint8_t * data; uint32_t alloc_size; hb_audio_resample_t *resample; }; static hb_buffer_t * Decode( hb_work_object_t * w ); static int declpcmInit( hb_work_object_t *, hb_job_t * ); static int declpcmWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** ); static void declpcmClose( hb_work_object_t * ); static int declpcmBSInfo( hb_work_object_t *, const hb_buffer_t *, hb_work_info_t * ); hb_work_object_t hb_declpcm = { WORK_DECLPCM, "LPCM decoder", declpcmInit, declpcmWork, declpcmClose, 0, declpcmBSInfo }; static const int hdr2samplerate[] = { 48000, 96000, 44100, 32000 }; static const int hdr2samplesize[] = { 16, 20, 24, 16 }; static const uint64_t hdr2layout[] = { AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_2_1, AV_CH_LAYOUT_QUAD, AV_CH_LAYOUT_5POINT0_BACK, AV_CH_LAYOUT_6POINT0_FRONT, AV_CH_LAYOUT_6POINT1, AV_CH_LAYOUT_7POINT1, }; static void lpcmInfo( hb_work_object_t *w, hb_buffer_t *in ) { hb_work_private_t * pv = w->private_data; /* * LPCM packets have a 7 byte header (the substream id is stripped off * before we get here so it's numbered -1 below):: * byte -1 Substream id * byte 0 Number of frames that begin in this packet * (last frame may finish in next packet) * byte 1,2 offset to first frame that begins in this packet (not including hdr) * byte 3: * bits 0-4 continuity counter (increments modulo 20) * bit 5 reserved * bit 6 audio mute on/off * bit 7 audio emphasis on/off * byte 4: * bits 0-2 #channels - 1 (e.g., stereo = 1) * bit 3 reserved * bits 4-5 sample rate (0=48K,1=96K,2=44.1K,3=32K) * bits 6-7 bits per sample (0=16 bit, 1=20 bit, 2=24 bit) * byte 5 Dynamic range control (0x80 = off) * * The audio is viewed as "frames" of 150 90KHz ticks each (80 samples @ 48KHz). * The frames are laid down continuously without regard to MPEG packet * boundaries. E.g., for 48KHz stereo, the first packet will contain 6 * frames plus the start of the 7th, the second packet will contain the * end of the 7th, 8-13 & the start of 14, etc. The frame structure is * important because the PTS on the packet gives the time of the first * frame that starts in the packet *NOT* the time of the first sample * in the packet. Also samples get split across packet boundaries * so we can't assume that we can consume all the data in one packet * on every call to the work routine. */ pv->offset = ( ( in->data[1] << 8 ) | in->data[2] ) + 2; if ( pv->offset >= HB_DVD_READ_BUFFER_SIZE ) { hb_log( "declpcm: illegal frame offset %d", pv->offset ); pv->offset = 2; /*XXX*/ } pv->nchannels = ( in->data[4] & 7 ) + 1; pv->samplerate = hdr2samplerate[ ( in->data[4] >> 4 ) & 0x3 ]; pv->sample_size = hdr2samplesize[in->data[4] >> 6]; // 20 and 24 bit lpcm is always encoded in sample pairs. So take this // into account when computing sizes. int chunk_size = pv->sample_size / 8; int samples_per_chunk = 1; switch( pv->sample_size ) { case 20: chunk_size = 5; samples_per_chunk = 2; break; case 24: chunk_size = 6; samples_per_chunk = 2; break; } /* * PCM frames have a constant duration (150 90KHz ticks). * We need to convert that to the amount of data expected. It's the * duration divided by the sample rate (to get #samples) times the number * of channels times the bits per sample divided by 8 to get bytes. * (we have to compute in bits because 20 bit samples are not an integral * number of bytes). We do all the multiplies first then the divides to * avoid truncation errors. */ /* * Don't trust the number of frames given in the header. We've seen * streams for which this is incorrect, and it can be computed. * pv->duration = in->data[0] * 150; */ int chunks = ( in->size - pv->offset ) / chunk_size; int samples = chunks * samples_per_chunk; // Calculate number of frames that start in this packet int frames = ( 90000 * samples / ( pv->samplerate * pv->nchannels ) + 149 ) / 150; pv->duration = frames * 150; pv->nchunks = ( pv->duration * pv->nchannels * pv->samplerate + samples_per_chunk - 1 ) / ( 90000 * samples_per_chunk ); pv->nsamples = ( pv->duration * pv->samplerate ) / 90000; pv->size = pv->nchunks * chunk_size; if (in->s.start != AV_NOPTS_VALUE) { pv->next_pts = in->s.start; } pv->scr_sequence = in->s.scr_sequence; } static int declpcmInit( hb_work_object_t * w, hb_job_t * job ) { hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) ); w->private_data = pv; pv->job = job; pv->next_pts = (int64_t)AV_NOPTS_VALUE; // Currently, samplerate conversion is performed in sync.c // So set output samplerate to input samplerate // This should someday get reworked to be part of an audio filter pipleine. pv->resample = hb_audio_resample_init(AV_SAMPLE_FMT_FLT, w->audio->config.in.samplerate, w->audio->config.out.mixdown, w->audio->config.out.normalize_mix_level); if (pv->resample == NULL) { hb_error("declpcmInit: hb_audio_resample_init() failed"); return 1; } return 0; } /* * Convert DVD encapsulated LPCM to floating point PCM audio buffers. * The amount of audio in a PCM frame is always <= the amount that will fit * in a DVD block (2048 bytes) but the standard doesn't require that the audio * frames line up with the DVD frames. Since audio frame boundaries are unrelated * to DVD PES boundaries, this routine has to reconstruct then extract the audio * frames. Because of the arbitrary alignment, it can output zero, one or two buf's. */ static int declpcmWork( hb_work_object_t * w, hb_buffer_t ** buf_in, hb_buffer_t ** buf_out ) { hb_work_private_t * pv = w->private_data; hb_buffer_t *in = *buf_in; hb_buffer_t *buf = NULL; hb_buffer_list_t list; hb_buffer_list_clear(&list); if (in->s.flags & HB_BUF_FLAG_EOF) { /* EOF on input stream - send it downstream & say that we're done */ *buf_out = in; *buf_in = NULL; return HB_WORK_DONE; } // if we have a frame to finish, add enough data from this buf // to finish it if (pv->size) { memcpy(pv->frame + pv->pos, in->data + 6, pv->size - pv->pos); buf = Decode( w ); hb_buffer_list_append(&list, buf); } /* save the (rest of) data from this buf in our frame buffer */ lpcmInfo( w, in ); int off = pv->offset; int amt = in->size - off; pv->pos = amt; memcpy( pv->frame, in->data + off, amt ); if (amt >= pv->size) { buf = Decode( w ); hb_buffer_list_append(&list, buf); pv->size = 0; } *buf_out = hb_buffer_list_clear(&list); return HB_WORK_OK; } static hb_buffer_t *Decode( hb_work_object_t *w ) { hb_work_private_t *pv = w->private_data; hb_buffer_t *out; if (pv->nsamples == 0) return NULL; int size = pv->nsamples * pv->nchannels * sizeof( float ); if (pv->alloc_size != size) { pv->data = realloc( pv->data, size ); pv->alloc_size = size; } float *odat = (float *)pv->data; int count = pv->nchunks / pv->nchannels; switch( pv->sample_size ) { case 16: // 2 byte, big endian, signed (the right shift sign extends) { uint8_t *frm = pv->frame; while ( count-- ) { int cc; for( cc = 0; cc < pv->nchannels; cc++ ) { // Shifts below result in sign extension which gives // us proper signed values. The final division adjusts // the range to [-1.0 ... 1.0] *odat++ = (float)( ( (int)( frm[0] << 24 ) >> 16 ) | frm[1] ) / 32768.0; frm += 2; } } } break; case 20: { // There will always be 2 groups of samples. A group is // a collection of samples that spans all channels. // The data for the samples is split. The first 2 msb // bytes for all samples is encoded first, then the remaining // lsb bits are encoded. uint8_t *frm = pv->frame; while ( count-- ) { int gg, cc; int shift = 4; uint8_t *lsb = frm + 4 * pv->nchannels; for( gg = 0; gg < 2; gg++ ) { for( cc = 0; cc < pv->nchannels; cc++ ) { // Shifts below result in sign extension which gives // us proper signed values. The final division adjusts // the range to [-1.0 ... 1.0] *odat = (float)( ( (int)( frm[0] << 24 ) >> 12 ) | ( frm[1] << 4 ) | ( ( ( lsb[0] >> shift ) & 0x0f ) ) ) / (16. * 32768.0); odat++; lsb += !shift; shift ^= 4; frm += 2; } } frm = lsb; } } break; case 24: { // There will always be 2 groups of samples. A group is // a collection of samples that spans all channels. // The data for the samples is split. The first 2 msb // bytes for all samples is encoded first, then the remaining // lsb bits are encoded. uint8_t *frm = pv->frame; while ( count-- ) { int gg, cc; uint8_t *lsb = frm + 4 * pv->nchannels; for( gg = 0; gg < 2; gg++ ) { for( cc = 0; cc < pv->nchannels; cc++ ) { // Shifts below result in sign extension which gives // us proper signed values. The final division adjusts // the range to [-1.0 ... 1.0] *odat++ = (float)( ( (int)( frm[0] << 24 ) >> 8 ) | ( frm[1] << 8 ) | lsb[0] ) / (256. * 32768.0); frm += 2; lsb++; } } frm = lsb; } } break; } hb_audio_resample_set_channel_layout(pv->resample, hdr2layout[pv->nchannels - 1]); hb_audio_resample_set_sample_rate(pv->resample, pv->samplerate); if (hb_audio_resample_update(pv->resample)) { hb_log("declpcm: hb_audio_resample_update() failed"); return NULL; } out = hb_audio_resample(pv->resample, (const uint8_t **)&pv->data, pv->nsamples); if (out != NULL) { out->s.start = pv->next_pts; out->s.duration = pv->duration; if (pv->next_pts != (int64_t)AV_NOPTS_VALUE) { pv->next_pts += pv->duration; out->s.stop = pv->next_pts; } out->s.scr_sequence = pv->scr_sequence; } return out; } static void declpcmClose( hb_work_object_t * w ) { hb_work_private_t * pv = w->private_data; if ( pv ) { hb_audio_resample_free(pv->resample); free( pv->data ); free( pv ); w->private_data = 0; } } static int declpcmBSInfo( hb_work_object_t *w, const hb_buffer_t *b, hb_work_info_t *info ) { int nchannels = ( b->data[4] & 7 ) + 1; int sample_size = hdr2samplesize[b->data[4] >> 6]; int rate = hdr2samplerate[ ( b->data[4] >> 4 ) & 0x3 ]; int bitrate = rate * sample_size * nchannels; int64_t duration = b->data[0] * 150; memset( info, 0, sizeof(*info) ); info->name = "LPCM"; info->rate.num = rate; info->rate.den = 1; info->bitrate = bitrate; info->flags = ( b->data[3] << 16 ) | ( b->data[4] << 8 ) | b->data[5]; info->matrix_encoding = AV_MATRIX_ENCODING_NONE; info->channel_layout = hdr2layout[nchannels - 1]; info->channel_map = &hb_libav_chan_map; info->sample_bit_depth = sample_size; info->samples_per_frame = ( duration * rate ) / 90000; return 1; }