FFmpeg  4.4
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 #include "libavutil/mem_internal.h"
26 
27 #define BITSTREAM_READER_LE
28 #include "avcodec.h"
29 #include "celp_filters.h"
30 #include "get_bits.h"
31 #include "internal.h"
32 #include "lpc.h"
33 #include "ra288.h"
34 
35 #define MAX_BACKWARD_FILTER_ORDER 36
36 #define MAX_BACKWARD_FILTER_LEN 40
37 #define MAX_BACKWARD_FILTER_NONREC 35
38 
39 #define RA288_BLOCK_SIZE 5
40 #define RA288_BLOCKS_PER_FRAME 32
41 
42 typedef struct RA288Context {
43  void (*vector_fmul)(float *dst, const float *src0, const float *src1,
44  int len);
45  DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
46  DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
47 
48  /** speech data history (spec: SB).
49  * Its first 70 coefficients are updated only at backward filtering.
50  */
51  float sp_hist[111];
52 
53  /// speech part of the gain autocorrelation (spec: REXP)
54  float sp_rec[37];
55 
56  /** log-gain history (spec: SBLG).
57  * Its first 28 coefficients are updated only at backward filtering.
58  */
59  float gain_hist[38];
60 
61  /// recursive part of the gain autocorrelation (spec: REXPLG)
62  float gain_rec[11];
63 } RA288Context;
64 
66 {
67  RA288Context *ractx = avctx->priv_data;
68  AVFloatDSPContext *fdsp;
69 
70  avctx->channels = 1;
73 
74  if (avctx->block_align != 38) {
75  av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
76  return AVERROR_PATCHWELCOME;
77  }
78 
80  if (!fdsp)
81  return AVERROR(ENOMEM);
82  ractx->vector_fmul = fdsp->vector_fmul;
83  av_free(fdsp);
84 
85  return 0;
86 }
87 
88 static void convolve(float *tgt, const float *src, int len, int n)
89 {
90  for (; n >= 0; n--)
91  tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
92 
93 }
94 
95 static void decode(RA288Context *ractx, float gain, int cb_coef)
96 {
97  int i;
98  double sumsum;
99  float sum, buffer[5];
100  float *block = ractx->sp_hist + 70 + 36; // current block
101  float *gain_block = ractx->gain_hist + 28;
102 
103  memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
104 
105  /* block 46 of G.728 spec */
106  sum = 32.0;
107  for (i=0; i < 10; i++)
108  sum -= gain_block[9-i] * ractx->gain_lpc[i];
109 
110  /* block 47 of G.728 spec */
111  sum = av_clipf(sum, 0, 60);
112 
113  /* block 48 of G.728 spec */
114  /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
115  sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
116 
117  for (i=0; i < 5; i++)
118  buffer[i] = codetable[cb_coef][i] * sumsum;
119 
121 
122  sum = FFMAX(sum, 5.0 / (1<<24));
123 
124  /* shift and store */
125  memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
126 
127  gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
128 
130 }
131 
132 /**
133  * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
134  *
135  * @param order filter order
136  * @param n input length
137  * @param non_rec number of non-recursive samples
138  * @param out filter output
139  * @param hist pointer to the input history of the filter
140  * @param out pointer to the non-recursive part of the output
141  * @param out2 pointer to the recursive part of the output
142  * @param window pointer to the windowing function table
143  */
144 static void do_hybrid_window(RA288Context *ractx,
145  int order, int n, int non_rec, float *out,
146  float *hist, float *out2, const float *window)
147 {
148  int i;
149  float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
150  float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
154 
155  av_assert2(order>=0);
156 
157  ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
158 
159  convolve(buffer1, work + order , n , order);
160  convolve(buffer2, work + order + n, non_rec, order);
161 
162  for (i=0; i <= order; i++) {
163  out2[i] = out2[i] * 0.5625 + buffer1[i];
164  out [i] = out2[i] + buffer2[i];
165  }
166 
167  /* Multiply by the white noise correcting factor (WNCF). */
168  *out *= 257.0 / 256.0;
169 }
170 
171 /**
172  * Backward synthesis filter, find the LPC coefficients from past speech data.
173  */
174 static void backward_filter(RA288Context *ractx,
175  float *hist, float *rec, const float *window,
176  float *lpc, const float *tab,
177  int order, int n, int non_rec, int move_size)
178 {
180 
181  do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
182 
183  if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
184  ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
185 
186  memmove(hist, hist + n, move_size*sizeof(*hist));
187 }
188 
189 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
190  int *got_frame_ptr, AVPacket *avpkt)
191 {
192  AVFrame *frame = data;
193  const uint8_t *buf = avpkt->data;
194  int buf_size = avpkt->size;
195  float *out;
196  int i, ret;
197  RA288Context *ractx = avctx->priv_data;
198  GetBitContext gb;
199 
200  if (buf_size < avctx->block_align) {
201  av_log(avctx, AV_LOG_ERROR,
202  "Error! Input buffer is too small [%d<%d]\n",
203  buf_size, avctx->block_align);
204  return AVERROR_INVALIDDATA;
205  }
206 
207  ret = init_get_bits8(&gb, buf, avctx->block_align);
208  if (ret < 0)
209  return ret;
210 
211  /* get output buffer */
213  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
214  return ret;
215  out = (float *)frame->data[0];
216 
217  for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
218  float gain = amptable[get_bits(&gb, 3)];
219  int cb_coef = get_bits(&gb, 6 + (i&1));
220 
221  decode(ractx, gain, cb_coef);
222 
223  memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
225 
226  if ((i & 7) == 3) {
227  backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
228  ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
229 
230  backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
231  ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
232  }
233  }
234 
235  *got_frame_ptr = 1;
236 
237  return avctx->block_align;
238 }
239 
241  .name = "real_288",
242  .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
243  .type = AVMEDIA_TYPE_AUDIO,
244  .id = AV_CODEC_ID_RA_288,
245  .priv_data_size = sizeof(RA288Context),
249 };
#define av_cold
Definition: attributes.h:88
uint8_t
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
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
audio channel layout utility functions
#define FFMAX(a, b)
Definition: common.h:103
#define av_clipf
Definition: common.h:170
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1893
static AVFrame * frame
int8_t exp
Definition: eval.c:72
static SDL_Window * window
Definition: ffplay.c:366
bitstream reader API header.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
#define AV_CH_LAYOUT_MONO
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:104
@ AV_CODEC_ID_RA_288
Definition: codec_id.h:411
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:117
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
for(j=16;j >0;--j)
int i
Definition: input.c:407
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:124
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
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:166
#define FFALIGN(x, a)
Definition: macros.h:48
#define LOCAL_ALIGNED(a, t, v,...)
Definition: mem_internal.h:113
const char data[16]
Definition: mxf.c:142
AVCodec ff_ra_288_decoder
Definition: ra288.c:240
#define MAX_BACKWARD_FILTER_ORDER
Definition: ra288.c:35
static void convolve(float *tgt, const float *src, int len, int n)
Definition: ra288.c:88
static void decode(RA288Context *ractx, float gain, int cb_coef)
Definition: ra288.c:95
static int ra288_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: ra288.c:189
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:174
static av_cold int ra288_decode_init(AVCodecContext *avctx)
Definition: ra288.c:65
#define MAX_BACKWARD_FILTER_NONREC
Definition: ra288.c:37
#define RA288_BLOCK_SIZE
Definition: ra288.c:39
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:144
#define RA288_BLOCKS_PER_FRAME
Definition: ra288.c:40
#define MAX_BACKWARD_FILTER_LEN
Definition: ra288.c:36
static const float gain_bw_tab[FFALIGN(10, 16)]
gain bandwidth broadening table
Definition: ra288.h:144
static const int16_t codetable[128][5]
Definition: ra288.h:34
static const float syn_bw_tab[FFALIGN(36, 16)]
synthesis bandwidth broadening table
Definition: ra288.h:134
static const float amptable[8]
Definition: ra288.h:29
static const float gain_window[FFALIGN(38, 16)]
Definition: ra288.h:123
static const float syn_window[FFALIGN(111, 16)]
Definition: ra288.h:101
typedef void(RENAME(mix_any_func_type))
static char buffer[20]
Definition: seek.c:32
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
int channels
number of audio channels
Definition: avcodec.h:1197
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1233
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
float gain_hist[38]
log-gain history (spec: SBLG).
Definition: ra288.c:59
float gain_lpc[FFALIGN(10, 16)]
LPC coefficients for gain (spec: GB)
Definition: ra288.c:46
float sp_rec[37]
speech part of the gain autocorrelation (spec: REXP)
Definition: ra288.c:54
float gain_rec[11]
recursive part of the gain autocorrelation (spec: REXPLG)
Definition: ra288.c:62
float sp_lpc[FFALIGN(36, 16)]
LPC coefficients for speech data (spec: A)
Definition: ra288.c:45
float sp_hist[111]
speech data history (spec: SB).
Definition: ra288.c:51
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Definition: ra288.c:43
#define av_free(p)
#define av_log(a,...)
#define src1
Definition: h264pred.c:140
#define src0
Definition: h264pred.c:139
#define src
Definition: vp8dsp.c:255
static int16_t block[64]
Definition: dct.c:116
FILE * out
Definition: movenc.c:54
static const struct twinvq_data tab
else temp
Definition: vf_mcdeint.c:259
int len