| /* |
| * 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 "put_bits.h" |
| #include "dcaenc.h" |
| #include "dcadata.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, uint8_t *frame, |
| int buf_size, void *data) |
| { |
| int i, k, channel; |
| DCAContext *c = avctx->priv_data; |
| int16_t *samples = data; |
| int real_channel = 0; |
| |
| for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */ |
| for (channel = 0; channel < c->prim_channels + 1; channel++) { |
| /* 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 */ |
| real_channel = c->channel_order_tab[channel]; |
| if (real_channel >= 0) { |
| 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, frame); |
| |
| return c->frame_size; |
| } |
| |
| static int encode_init(AVCodecContext *avctx) |
| { |
| DCAContext *c = avctx->priv_data; |
| int i; |
| |
| c->prim_channels = avctx->channels; |
| c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6); |
| |
| switch (avctx->channel_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 (dca_sample_rates[i] && (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, ", 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 = CODEC_ID_DTS, |
| .priv_data_size = sizeof(DCAContext), |
| .init = encode_init, |
| .encode = encode_frame, |
| .capabilities = CODEC_CAP_EXPERIMENTAL, |
| .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, |
| }; |
