FFmpeg
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  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
250 };
AVCodec
AVCodec.
Definition: codec.h:197
FF_CODEC_CAP_INIT_THREADSAFE
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1149
mem_internal.h
out
FILE * out
Definition: movenc.c:54
codetable
static const int16_t codetable[128][5]
Definition: ra288.h:34
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:90
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
compute_lpc_coefs
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
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:365
data
const char data[16]
Definition: mxf.c:142
MAX_BACKWARD_FILTER_NONREC
#define MAX_BACKWARD_FILTER_NONREC
Definition: ra288.c:37
backward_filter
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
lpc.h
ff_celp_lp_synthesis_filterf
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
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
window
static SDL_Window * window
Definition: ffplay.c:366
decode
static void decode(RA288Context *ractx, float gain, int cb_coef)
Definition: ra288.c:95
GetBitContext
Definition: get_bits.h:61
tab
static const struct twinvq_data tab
Definition: twinvq_data.h:10345
MAX_BACKWARD_FILTER_ORDER
#define MAX_BACKWARD_FILTER_ORDER
Definition: ra288.c:35
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:581
gain_window
static const float gain_window[FFALIGN(38, 16)]
Definition: ra288.h:123
syn_bw_tab
static const float syn_bw_tab[FFALIGN(36, 16)]
synthesis bandwidth broadening table
Definition: ra288.h:134
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:181
av_cold
#define av_cold
Definition: attributes.h:90
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
LOCAL_ALIGNED
#define LOCAL_ALIGNED(a, t, v,...)
Definition: mem_internal.h:113
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
do_hybrid_window
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
get_bits.h
gain_bw_tab
static const float gain_bw_tab[FFALIGN(10, 16)]
gain bandwidth broadening table
Definition: ra288.h:144
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
av_clipf
#define av_clipf
Definition: common.h:170
work
must be printed separately If there s no standard function for printing the type you the WRITE_1D_FUNC_ARGV macro is a very quick way to create one See libavcodec dv_tablegen c for an example The h file This file should the initialization functions should not do and instead of the variable declarations the generated *_tables h file should be included Since that will be generated in the build the path must be i e not Makefile changes To make the automatic table creation work
Definition: tablegen.txt:66
src
#define src
Definition: vp8dsp.c:255
RA288_BLOCKS_PER_FRAME
#define RA288_BLOCKS_PER_FRAME
Definition: ra288.c:40
RA288Context::sp_lpc
float sp_lpc[FFALIGN(36, 16)]
LPC coefficients for speech data (spec: A)
Definition: ra288.c:45
celp_filters.h
exp
int8_t exp
Definition: eval.c:72
ra288_decode_init
static av_cold int ra288_decode_init(AVCodecContext *avctx)
Definition: ra288.c:65
float_dsp.h
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
AV_CODEC_CAP_CHANNEL_CONF
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:104
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1638
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:366
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
AVFloatDSPContext::vector_fmul
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
FFMAX
#define FFMAX(a, b)
Definition: common.h:103
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1106
AVFloatDSPContext
Definition: float_dsp.h:24
convolve
static void convolve(float *tgt, const float *src, int len, int n)
Definition: ra288.c:88
RA288Context
Definition: ra288.c:42
src0
#define src0
Definition: h264pred.c:139
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1099
RA288_BLOCK_SIZE
#define RA288_BLOCK_SIZE
Definition: ra288.c:39
DECLARE_ALIGNED
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:117
src1
#define src1
Definition: h264pred.c:140
av_assert2
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
AV_CODEC_ID_RA_288
@ AV_CODEC_ID_RA_288
Definition: codec_id.h:409
i
int i
Definition: input.c:407
internal.h
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:204
RA288Context::gain_lpc
float gain_lpc[FFALIGN(10, 16)]
LPC coefficients for gain (spec: GB)
Definition: ra288.c:46
len
int len
Definition: vorbis_enc_data.h:452
syn_window
static const float syn_window[FFALIGN(111, 16)]
Definition: ra288.h:101
RA288Context::gain_hist
float gain_hist[38]
log-gain history (spec: SBLG).
Definition: ra288.c:59
avcodec.h
RA288Context::vector_fmul
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Definition: ra288.c:43
ret
ret
Definition: filter_design.txt:187
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1135
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
RA288Context::sp_rec
float sp_rec[37]
speech part of the gain autocorrelation (spec: REXP)
Definition: ra288.c:54
RA288Context::gain_rec
float gain_rec[11]
recursive part of the gain autocorrelation (spec: REXPLG)
Definition: ra288.c:62
ra288_decode_frame
static int ra288_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: ra288.c:189
AVCodecContext
main external API structure.
Definition: avcodec.h:501
channel_layout.h
buffer
the frame and frame reference mechanism is intended to as much as expensive copies of that data while still allowing the filters to produce correct results The data is stored in buffers represented by AVFrame structures Several references can point to the same frame buffer
Definition: filter_design.txt:49
temp
else temp
Definition: vf_mcdeint.c:259
avpriv_scalarproduct_float_c
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_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:310
amptable
static const float amptable[8]
Definition: ra288.h:29
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
AVPacket
This structure stores compressed data.
Definition: packet.h:342
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:528
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
block
The exact code depends on how similar the blocks are and how related they are to the block
Definition: filter_design.txt:207
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
ff_ra_288_decoder
const AVCodec ff_ra_288_decoder
Definition: ra288.c:240
RA288Context::sp_hist
float sp_hist[111]
speech data history (spec: SB).
Definition: ra288.c:51
MAX_BACKWARD_FILTER_LEN
#define MAX_BACKWARD_FILTER_LEN
Definition: ra288.c:36
ra288.h
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63