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ra288.c
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1 /*
2  * RealAudio 2.0 (28.8K)
3  * Copyright (c) 2003 The FFmpeg project
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
23 #include "libavutil/float_dsp.h"
24 #include "libavutil/internal.h"
25 
26 #define BITSTREAM_READER_LE
27 #include "avcodec.h"
28 #include "celp_filters.h"
29 #include "get_bits.h"
30 #include "internal.h"
31 #include "lpc.h"
32 #include "ra288.h"
33 
34 #define MAX_BACKWARD_FILTER_ORDER 36
35 #define MAX_BACKWARD_FILTER_LEN 40
36 #define MAX_BACKWARD_FILTER_NONREC 35
37 
38 #define RA288_BLOCK_SIZE 5
39 #define RA288_BLOCKS_PER_FRAME 32
40 
41 typedef struct RA288Context {
43  DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
44  DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
45 
46  /** speech data history (spec: SB).
47  * Its first 70 coefficients are updated only at backward filtering.
48  */
49  float sp_hist[111];
50 
51  /// speech part of the gain autocorrelation (spec: REXP)
52  float sp_rec[37];
53 
54  /** log-gain history (spec: SBLG).
55  * Its first 28 coefficients are updated only at backward filtering.
56  */
57  float gain_hist[38];
58 
59  /// recursive part of the gain autocorrelation (spec: REXPLG)
60  float gain_rec[11];
61 } RA288Context;
62 
64 {
65  RA288Context *ractx = avctx->priv_data;
66 
67  av_freep(&ractx->fdsp);
68 
69  return 0;
70 }
71 
73 {
74  RA288Context *ractx = avctx->priv_data;
75 
76  avctx->channels = 1;
79 
80  if (avctx->block_align <= 0) {
81  av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
82  return AVERROR_PATCHWELCOME;
83  }
84 
86  if (!ractx->fdsp)
87  return AVERROR(ENOMEM);
88 
89  return 0;
90 }
91 
92 static void convolve(float *tgt, const float *src, int len, int n)
93 {
94  for (; n >= 0; n--)
95  tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
96 
97 }
98 
99 static void decode(RA288Context *ractx, float gain, int cb_coef)
100 {
101  int i;
102  double sumsum;
103  float sum, buffer[5];
104  float *block = ractx->sp_hist + 70 + 36; // current block
105  float *gain_block = ractx->gain_hist + 28;
106 
107  memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
108 
109  /* block 46 of G.728 spec */
110  sum = 32.0;
111  for (i=0; i < 10; i++)
112  sum -= gain_block[9-i] * ractx->gain_lpc[i];
113 
114  /* block 47 of G.728 spec */
115  sum = av_clipf(sum, 0, 60);
116 
117  /* block 48 of G.728 spec */
118  /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
119  sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
120 
121  for (i=0; i < 5; i++)
122  buffer[i] = codetable[cb_coef][i] * sumsum;
123 
124  sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
125 
126  sum = FFMAX(sum, 5.0 / (1<<24));
127 
128  /* shift and store */
129  memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
130 
131  gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
132 
133  ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
134 }
135 
136 /**
137  * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
138  *
139  * @param order filter order
140  * @param n input length
141  * @param non_rec number of non-recursive samples
142  * @param out filter output
143  * @param hist pointer to the input history of the filter
144  * @param out pointer to the non-recursive part of the output
145  * @param out2 pointer to the recursive part of the output
146  * @param window pointer to the windowing function table
147  */
148 static void do_hybrid_window(RA288Context *ractx,
149  int order, int n, int non_rec, float *out,
150  float *hist, float *out2, const float *window)
151 {
152  int i;
153  float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
154  float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
158 
159  av_assert2(order>=0);
160 
161  ractx->fdsp->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
162 
163  convolve(buffer1, work + order , n , order);
164  convolve(buffer2, work + order + n, non_rec, order);
165 
166  for (i=0; i <= order; i++) {
167  out2[i] = out2[i] * 0.5625 + buffer1[i];
168  out [i] = out2[i] + buffer2[i];
169  }
170 
171  /* Multiply by the white noise correcting factor (WNCF). */
172  *out *= 257.0 / 256.0;
173 }
174 
175 /**
176  * Backward synthesis filter, find the LPC coefficients from past speech data.
177  */
178 static void backward_filter(RA288Context *ractx,
179  float *hist, float *rec, const float *window,
180  float *lpc, const float *tab,
181  int order, int n, int non_rec, int move_size)
182 {
184 
185  do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
186 
187  if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
188  ractx->fdsp->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
189 
190  memmove(hist, hist + n, move_size*sizeof(*hist));
191 }
192 
193 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
194  int *got_frame_ptr, AVPacket *avpkt)
195 {
196  AVFrame *frame = data;
197  const uint8_t *buf = avpkt->data;
198  int buf_size = avpkt->size;
199  float *out;
200  int i, ret;
201  RA288Context *ractx = avctx->priv_data;
202  GetBitContext gb;
203 
204  if (buf_size < avctx->block_align) {
205  av_log(avctx, AV_LOG_ERROR,
206  "Error! Input buffer is too small [%d<%d]\n",
207  buf_size, avctx->block_align);
208  return AVERROR_INVALIDDATA;
209  }
210 
211  ret = init_get_bits8(&gb, buf, avctx->block_align);
212  if (ret < 0)
213  return ret;
214 
215  /* get output buffer */
217  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
218  return ret;
219  out = (float *)frame->data[0];
220 
221  for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
222  float gain = amptable[get_bits(&gb, 3)];
223  int cb_coef = get_bits(&gb, 6 + (i&1));
224 
225  decode(ractx, gain, cb_coef);
226 
227  memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
228  out += RA288_BLOCK_SIZE;
229 
230  if ((i & 7) == 3) {
231  backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
232  ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
233 
234  backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
235  ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
236  }
237  }
238 
239  *got_frame_ptr = 1;
240 
241  return avctx->block_align;
242 }
243 
245  .name = "real_288",
246  .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
247  .type = AVMEDIA_TYPE_AUDIO,
248  .id = AV_CODEC_ID_RA_288,
249  .priv_data_size = sizeof(RA288Context),
252  .close = ra288_decode_close,
253  .capabilities = AV_CODEC_CAP_DR1,
254 };
float sp_lpc[FFALIGN(36, 16)]
LPC coefficients for speech data (spec: A)
Definition: ra288.c:43
static av_cold int ra288_decode_close(AVCodecContext *avctx)
Definition: ra288.c:63
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
Definition: celp_filters.c:84
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
float gain_lpc[FFALIGN(10, 16)]
LPC coefficients for gain (spec: GB)
Definition: ra288.c:44
static void backward_filter(RA288Context *ractx, float *hist, float *rec, const float *window, float *lpc, const float *tab, int order, int n, int non_rec, int move_size)
Backward synthesis filter, find the LPC coefficients from past speech data.
Definition: ra288.c:178
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:247
else temp
Definition: vf_mcdeint.c:259
#define MAX_BACKWARD_FILTER_ORDER
Definition: ra288.c:34
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
int size
Definition: avcodec.h:1589
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:53
static void decode(RA288Context *ractx, float gain, int cb_coef)
Definition: ra288.c:99
AVCodec.
Definition: avcodec.h:3559
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2455
static const float amptable[8]
Definition: ra288.h:28
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:108
float sp_hist[111]
speech data history (spec: SB).
Definition: ra288.c:49
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2426
uint8_t
#define av_cold
Definition: attributes.h:82
#define MAX_BACKWARD_FILTER_NONREC
Definition: ra288.c:36
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:63
float gain_hist[38]
log-gain history (spec: SBLG).
Definition: ra288.c:57
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1588
bitstream reader API header.
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats...
Definition: float_dsp.h:38
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
#define MAX_BACKWARD_FILTER_LEN
Definition: ra288.c:35
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1752
const char * name
Name of the codec implementation.
Definition: avcodec.h:3566
static void convolve(float *tgt, const float *src, int len, int n)
Definition: ra288.c:92
#define FFMAX(a, b)
Definition: common.h:94
int8_t exp
Definition: eval.c:64
static const float syn_window[FFALIGN(111, 16)]
Definition: ra288.h:100
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2469
AVCodec ff_ra_288_decoder
Definition: ra288.c:244
common internal API header
audio channel layout utility functions
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:883
#define RA288_BLOCKS_PER_FRAME
Definition: ra288.c:39
static const float gain_window[FFALIGN(38, 16)]
Definition: ra288.h:122
int n
Definition: avisynth_c.h:547
static int AAC_RENAME() compute_lpc_coefs(const LPC_TYPE *autoc, int max_order, LPC_TYPE *lpc, int lpc_stride, int fail, int normalize)
Levinson-Durbin recursion.
Definition: lpc.h:163
#define src
Definition: vp9dsp.c:530
static const float syn_bw_tab[FFALIGN(36, 16)]
synthesis bandwidth broadening table
Definition: ra288.h:133
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
AVFloatDSPContext * fdsp
Definition: ra288.c:42
static const int16_t codetable[128][5]
Definition: ra288.h:33
Libavcodec external API header.
float sp_rec[37]
speech part of the gain autocorrelation (spec: REXP)
Definition: ra288.c:52
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:437
main external API structure.
Definition: avcodec.h:1657
static void do_hybrid_window(RA288Context *ractx, int order, int n, int non_rec, float *out, float *hist, float *out2, const float *window)
Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
Definition: ra288.c:148
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:928
void * buf
Definition: avisynth_c.h:553
static const float gain_bw_tab[FFALIGN(10, 16)]
gain bandwidth broadening table
Definition: ra288.h:143
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
#define RA288_BLOCK_SIZE
Definition: ra288.c:38
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:198
static av_cold int ra288_decode_init(AVCodecContext *avctx)
Definition: ra288.c:72
static int ra288_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: ra288.c:193
common internal api header.
#define LOCAL_ALIGNED(a, t, v,...)
Definition: internal.h:110
float gain_rec[11]
recursive part of the gain autocorrelation (spec: REXPLG)
Definition: ra288.c:60
void * priv_data
Definition: avcodec.h:1699
int len
int channels
number of audio channels
Definition: avcodec.h:2419
static const struct twinvq_data tab
FILE * out
Definition: movenc.c:54
#define av_freep(p)
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
Definition: avcodec.h:1565
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:956
for(j=16;j >0;--j)
GLuint buffer
Definition: opengl_enc.c:102
static int16_t block[64]
Definition: dct-test.c:112