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Diffstat (limited to 'libavcodec/dcaenc.c')
| -rw-r--r-- | libavcodec/dcaenc.c | 603 |
1 files changed, 603 insertions, 0 deletions
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c new file mode 100644 index 0000000000..104fd87cda --- /dev/null +++ b/libavcodec/dcaenc.c @@ -0,0 +1,603 @@ +/* + * DCA encoder + * Copyright (C) 2008 Alexander E. Patrakov + * 2010 Benjamin Larsson + * 2011 Xiang Wang + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/common.h" +#include "libavutil/avassert.h" +#include "libavutil/audioconvert.h" +#include "avcodec.h" +#include "get_bits.h" +#include "internal.h" +#include "put_bits.h" +#include "dcaenc.h" +#include "dcadata.h" +#include "dca.h" + +#undef NDEBUG + +#define MAX_CHANNELS 6 +#define DCA_SUBBANDS_32 32 +#define DCA_MAX_FRAME_SIZE 16383 +#define DCA_HEADER_SIZE 13 + +#define DCA_SUBBANDS 32 ///< Subband activity count +#define QUANTIZER_BITS 16 +#define SUBFRAMES 1 +#define SUBSUBFRAMES 4 +#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8) +#define LFE_BITS 8 +#define LFE_INTERPOLATION 64 +#define LFE_PRESENT 2 +#define LFE_MISSING 0 + +static const int8_t dca_lfe_index[] = { + 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3 +}; + +static const int8_t dca_channel_reorder_lfe[][9] = { + { 0, -1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 1, 2, 0, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, 2, -1, -1, -1, -1, -1 }, + { 1, 2, 0, -1, 3, -1, -1, -1, -1 }, + { 0, 1, -1, 2, 3, -1, -1, -1, -1 }, + { 1, 2, 0, -1, 3, 4, -1, -1, -1 }, + { 2, 3, -1, 0, 1, 4, 5, -1, -1 }, + { 1, 2, 0, -1, 3, 4, 5, -1, -1 }, + { 0, -1, 4, 5, 2, 3, 1, -1, -1 }, + { 3, 4, 1, -1, 0, 2, 5, 6, -1 }, + { 2, 3, -1, 5, 7, 0, 1, 4, 6 }, + { 3, 4, 1, -1, 0, 2, 5, 7, 6 }, +}; + +static const int8_t dca_channel_reorder_nolfe[][9] = { + { 0, -1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 0, 1, -1, -1, -1, -1, -1, -1, -1 }, + { 1, 2, 0, -1, -1, -1, -1, -1, -1 }, + { 0, 1, 2, -1, -1, -1, -1, -1, -1 }, + { 1, 2, 0, 3, -1, -1, -1, -1, -1 }, + { 0, 1, 2, 3, -1, -1, -1, -1, -1 }, + { 1, 2, 0, 3, 4, -1, -1, -1, -1 }, + { 2, 3, 0, 1, 4, 5, -1, -1, -1 }, + { 1, 2, 0, 3, 4, 5, -1, -1, -1 }, + { 0, 4, 5, 2, 3, 1, -1, -1, -1 }, + { 3, 4, 1, 0, 2, 5, 6, -1, -1 }, + { 2, 3, 5, 7, 0, 1, 4, 6, -1 }, + { 3, 4, 1, 0, 2, 5, 7, 6, -1 }, +}; + +typedef struct { + PutBitContext pb; + int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */ + int start[MAX_CHANNELS]; + int frame_size; + int prim_channels; + int lfe_channel; + int sample_rate_code; + int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32]; + int lfe_scale_factor; + int lfe_data[SUBFRAMES*SUBSUBFRAMES*4]; + + int a_mode; ///< audio channels arrangement + int num_channel; + int lfe_state; + int lfe_offset; + const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe + + int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)]; + int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */ +} DCAContext; + +static int32_t cos_table[128]; + +static inline int32_t mul32(int32_t a, int32_t b) +{ + int64_t r = (int64_t) a * b; + /* round the result before truncating - improves accuracy */ + return (r + 0x80000000) >> 32; +} + +/* Integer version of the cosine modulated Pseudo QMF */ + +static void qmf_init(void) +{ + int i; + int32_t c[17], s[17]; + s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */ + c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */ + + for (i = 1; i <= 16; i++) { + s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908)); + c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028)); + } + + for (i = 0; i < 16; i++) { + cos_table[i ] = c[i] >> 3; /* avoid output overflow */ + cos_table[i + 16] = s[16 - i] >> 3; + cos_table[i + 32] = -s[i] >> 3; + cos_table[i + 48] = -c[16 - i] >> 3; + cos_table[i + 64] = -c[i] >> 3; + cos_table[i + 80] = -s[16 - i] >> 3; + cos_table[i + 96] = s[i] >> 3; + cos_table[i + 112] = c[16 - i] >> 3; + } +} + +static int32_t band_delta_factor(int band, int sample_num) +{ + int index = band * (2 * sample_num + 1); + if (band == 0) + return 0x07ffffff; + else + return cos_table[index & 127]; +} + +static void add_new_samples(DCAContext *c, const int32_t *in, + int count, int channel) +{ + int i; + + /* Place new samples into the history buffer */ + for (i = 0; i < count; i++) { + c->history[channel][c->start[channel] + i] = in[i]; + av_assert0(c->start[channel] + i < 512); + } + c->start[channel] += count; + if (c->start[channel] == 512) + c->start[channel] = 0; + av_assert0(c->start[channel] < 512); +} + +static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32], + int channel) +{ + int band, i, j, k; + int32_t resp; + int32_t accum[DCA_SUBBANDS_32] = {0}; + + add_new_samples(c, in, DCA_SUBBANDS_32, channel); + + /* Calculate the dot product of the signal with the (possibly inverted) + reference decoder's response to this vector: + (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0) + so that -1.0 cancels 1.0 from the previous step */ + + for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++) + accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]); + for (i = 0; i < c->start[channel]; k++, j++, i++) + accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]); + + resp = 0; + /* TODO: implement FFT instead of this naive calculation */ + for (band = 0; band < DCA_SUBBANDS_32; band++) { + for (j = 0; j < 32; j++) + resp += mul32(accum[j], band_delta_factor(band, j)); + + out[band] = (band & 2) ? (-resp) : resp; + } +} + +static int32_t lfe_fir_64i[512]; +static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION]) +{ + int i, j; + int channel = c->prim_channels; + int32_t accum = 0; + + add_new_samples(c, in, LFE_INTERPOLATION, channel); + for (i = c->start[channel], j = 0; i < 512; i++, j++) + accum += mul32(c->history[channel][i], lfe_fir_64i[j]); + for (i = 0; i < c->start[channel]; i++, j++) + accum += mul32(c->history[channel][i], lfe_fir_64i[j]); + return accum; +} + +static void init_lfe_fir(void) +{ + static int initialized = 0; + int i; + if (initialized) + return; + + for (i = 0; i < 512; i++) + lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t + initialized = 1; +} + +static void put_frame_header(DCAContext *c) +{ + /* SYNC */ + put_bits(&c->pb, 16, 0x7ffe); + put_bits(&c->pb, 16, 0x8001); + + /* Frame type: normal */ + put_bits(&c->pb, 1, 1); + + /* Deficit sample count: none */ + put_bits(&c->pb, 5, 31); + + /* CRC is not present */ + put_bits(&c->pb, 1, 0); + + /* Number of PCM sample blocks */ + put_bits(&c->pb, 7, PCM_SAMPLES-1); + + /* Primary frame byte size */ + put_bits(&c->pb, 14, c->frame_size-1); + + /* Audio channel arrangement: L + R (stereo) */ + put_bits(&c->pb, 6, c->num_channel); + + /* Core audio sampling frequency */ + put_bits(&c->pb, 4, c->sample_rate_code); + + /* Transmission bit rate: 1411.2 kbps */ + put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */ + + /* Embedded down mix: disabled */ + put_bits(&c->pb, 1, 0); + + /* Embedded dynamic range flag: not present */ + put_bits(&c->pb, 1, 0); + + /* Embedded time stamp flag: not present */ + put_bits(&c->pb, 1, 0); + + /* Auxiliary data flag: not present */ + put_bits(&c->pb, 1, 0); + + /* HDCD source: no */ + put_bits(&c->pb, 1, 0); + + /* Extension audio ID: N/A */ + put_bits(&c->pb, 3, 0); + + /* Extended audio data: not present */ + put_bits(&c->pb, 1, 0); + + /* Audio sync word insertion flag: after each sub-frame */ + put_bits(&c->pb, 1, 0); + + /* Low frequency effects flag: not present or interpolation factor=64 */ + put_bits(&c->pb, 2, c->lfe_state); + + /* Predictor history switch flag: on */ + put_bits(&c->pb, 1, 1); + + /* No CRC */ + /* Multirate interpolator switch: non-perfect reconstruction */ + put_bits(&c->pb, 1, 0); + + /* Encoder software revision: 7 */ + put_bits(&c->pb, 4, 7); + + /* Copy history: 0 */ + put_bits(&c->pb, 2, 0); + + /* Source PCM resolution: 16 bits, not DTS ES */ + put_bits(&c->pb, 3, 0); + + /* Front sum/difference coding: no */ + put_bits(&c->pb, 1, 0); + + /* Surrounds sum/difference coding: no */ + put_bits(&c->pb, 1, 0); + + /* Dialog normalization: 0 dB */ + put_bits(&c->pb, 4, 0); +} + +static void put_primary_audio_header(DCAContext *c) +{ + static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; + static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; + + int ch, i; + /* Number of subframes */ + put_bits(&c->pb, 4, SUBFRAMES - 1); + + /* Number of primary audio channels */ + put_bits(&c->pb, 3, c->prim_channels - 1); + + /* Subband activity count */ + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, 5, DCA_SUBBANDS - 2); + + /* High frequency VQ start subband */ + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, 5, DCA_SUBBANDS - 1); + + /* Joint intensity coding index: 0, 0 */ + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, 3, 0); + + /* Transient mode codebook: A4, A4 (arbitrary) */ + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, 2, 0); + + /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */ + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, 3, 6); + + /* Bit allocation quantizer select: linear 5-bit */ + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, 3, 6); + + /* Quantization index codebook select: dummy data + to avoid transmission of scale factor adjustment */ + + for (i = 1; i < 11; i++) + for (ch = 0; ch < c->prim_channels; ch++) + put_bits(&c->pb, bitlen[i], thr[i]); + + /* Scale factor adjustment index: not transmitted */ +} + +/** + * 8-23 bits quantization + * @param sample + * @param bits + */ +static inline uint32_t quantize(int32_t sample, int bits) +{ + av_assert0(sample < 1 << (bits - 1)); + av_assert0(sample >= -(1 << (bits - 1))); + return sample & ((1 << bits) - 1); +} + +static inline int find_scale_factor7(int64_t max_value, int bits) +{ + int i = 0, j = 128, q; + max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1); + while (i < j) { + q = (i + j) >> 1; + if (max_value < scale_factor_quant7[q]) + j = q; + else + i = q + 1; + } + av_assert1(i < 128); + return i; +} + +static inline void put_sample7(DCAContext *c, int64_t sample, int bits, + int scale_factor) +{ + sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]); + put_bits(&c->pb, bits, quantize((int) sample, bits)); +} + +static void put_subframe(DCAContext *c, + int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32], + int subframe) +{ + int i, sub, ss, ch, max_value; + int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe; + + /* Subsubframes count */ + put_bits(&c->pb, 2, SUBSUBFRAMES -1); + + /* Partial subsubframe sample count: dummy */ + put_bits(&c->pb, 3, 0); + + /* Prediction mode: no ADPCM, in each channel and subband */ + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) + put_bits(&c->pb, 1, 0); + + /* Prediction VQ addres: not transmitted */ + /* Bit allocation index */ + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) + put_bits(&c->pb, 5, QUANTIZER_BITS+3); + + if (SUBSUBFRAMES > 1) { + /* Transition mode: none for each channel and subband */ + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) + put_bits(&c->pb, 1, 0); /* codebook A4 */ + } + + /* Determine scale_factor */ + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) { + max_value = 0; + for (i = 0; i < 8 * SUBSUBFRAMES; i++) + max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub])); + c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS); + } + + if (c->lfe_channel) { + max_value = 0; + for (i = 0; i < 4 * SUBSUBFRAMES; i++) + max_value = FFMAX(max_value, FFABS(lfe_data[i])); + c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS); + } + + /* Scale factors: the same for each channel and subband, + encoded according to Table D.1.2 */ + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) + put_bits(&c->pb, 7, c->scale_factor[ch][sub]); + + /* Joint subband scale factor codebook select: not transmitted */ + /* Scale factors for joint subband coding: not transmitted */ + /* Stereo down-mix coefficients: not transmitted */ + /* Dynamic range coefficient: not transmitted */ + /* Stde information CRC check word: not transmitted */ + /* VQ encoded high frequency subbands: not transmitted */ + + /* LFE data */ + if (c->lfe_channel) { + for (i = 0; i < 4 * SUBSUBFRAMES; i++) + put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor); + put_bits(&c->pb, 8, c->lfe_scale_factor); + } + + /* Audio data (subsubframes) */ + + for (ss = 0; ss < SUBSUBFRAMES ; ss++) + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) + for (i = 0; i < 8; i++) + put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]); + + /* DSYNC */ + put_bits(&c->pb, 16, 0xffff); +} + +static void put_frame(DCAContext *c, + int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32], + uint8_t *frame) +{ + int i; + init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE); + + put_primary_audio_header(c); + for (i = 0; i < SUBFRAMES; i++) + put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i); + + flush_put_bits(&c->pb); + c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE; + + init_put_bits(&c->pb, frame, DCA_HEADER_SIZE); + put_frame_header(c); + flush_put_bits(&c->pb); +} + +static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + int i, k, channel; + DCAContext *c = avctx->priv_data; + const int16_t *samples; + int ret, real_channel = 0; + + if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE))) + return ret; + + samples = (const int16_t *)frame->data[0]; + for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */ + for (channel = 0; channel < c->prim_channels + 1; channel++) { + real_channel = c->channel_order_tab[channel]; + if (real_channel >= 0) { + /* Get 32 PCM samples */ + for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */ + c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16; + } + /* Put subband samples into the proper place */ + qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel); + } + } + } + + if (c->lfe_channel) { + for (i = 0; i < PCM_SAMPLES / 2; i++) { + for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */ + c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16; + c->lfe_data[i] = lfe_downsample(c, c->pcm); + } + } + + put_frame(c, c->subband, avpkt->data); + + avpkt->size = c->frame_size; + *got_packet_ptr = 1; + return 0; +} + +static int encode_init(AVCodecContext *avctx) +{ + DCAContext *c = avctx->priv_data; + int i; + uint64_t layout = avctx->channel_layout; + + c->prim_channels = avctx->channels; + c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6); + + if (!layout) { + av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The " + "encoder will guess the layout, but it " + "might be incorrect.\n"); + layout = av_get_default_channel_layout(avctx->channels); + } + switch (layout) { + case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break; + case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break; + case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break; + case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break; + case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break; + default: + av_log(avctx, AV_LOG_ERROR, + "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n"); + return AVERROR_PATCHWELCOME; + } + + if (c->lfe_channel) { + init_lfe_fir(); + c->prim_channels--; + c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode]; + c->lfe_state = LFE_PRESENT; + c->lfe_offset = dca_lfe_index[c->a_mode]; + } else { + c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode]; + c->lfe_state = LFE_MISSING; + } + + for (i = 0; i < 16; i++) { + if (avpriv_dca_sample_rates[i] && (avpriv_dca_sample_rates[i] == avctx->sample_rate)) + break; + } + if (i == 16) { + av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate); + for (i = 0; i < 16; i++) + av_log(avctx, AV_LOG_ERROR, "%d, ", avpriv_dca_sample_rates[i]); + av_log(avctx, AV_LOG_ERROR, "supported.\n"); + return -1; + } + c->sample_rate_code = i; + + avctx->frame_size = 32 * PCM_SAMPLES; + + if (!cos_table[127]) + qmf_init(); + return 0; +} + +AVCodec ff_dca_encoder = { + .name = "dca", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_DTS, + .priv_data_size = sizeof(DCAContext), + .init = encode_init, + .encode2 = encode_frame, + .capabilities = CODEC_CAP_EXPERIMENTAL, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), +}; |