FFmpeg: libavcodec/ra288.c Source File

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ra288.c

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1 /*

2 * RealAudio 2.0 (28.8K)

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 */

22 #include "libavutil/channel_layout.h"

23 #include "libavutil/float_dsp.h"

24 #include "libavutil/internal.h"

25 #include "avcodec.h"

26 #include "internal.h"

27 #define BITSTREAM_READER_LE

28 #include "get_bits.h"

29 #include "ra288.h"

30 #include "lpc.h"

31 #include "celp_filters.h"

33 #define MAX_BACKWARD_FILTER_ORDER 36

34 #define MAX_BACKWARD_FILTER_LEN 40

35 #define MAX_BACKWARD_FILTER_NONREC 35

37 #define RA288_BLOCK_SIZE 5

38 #define RA288_BLOCKS_PER_FRAME 32

40 typedef struct RA288Context {

41 AVFloatDSPContext *fdsp;

42 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)

43 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)

45 /** speech data history (spec: SB).

46 * Its first 70 coefficients are updated only at backward filtering.

47 */

48 float sp_hist[111];

50 /// speech part of the gain autocorrelation (spec: REXP)

51 float sp_rec[37];

53 /** log-gain history (spec: SBLG).

54 * Its first 28 coefficients are updated only at backward filtering.

55 */

56 float gain_hist[38];

58 /// recursive part of the gain autocorrelation (spec: REXPLG)

59 float gain_rec[11];

60 } RA288Context;

62 static av_cold int ra288_decode_close(AVCodecContext *avctx)

63 {

64 RA288Context *ractx = avctx->priv_data;

66 av_freep(&ractx->fdsp);

68 return 0;

69 }

71 static av_cold int ra288_decode_init(AVCodecContext *avctx)

72 {

73 RA288Context *ractx = avctx->priv_data;

75 avctx->channels = 1;

76 avctx->channel_layout = AV_CH_LAYOUT_MONO;

77 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

79 if (avctx->block_align <= 0) {

80 av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");

81 return AVERROR_PATCHWELCOME;

82 }

84 ractx->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);

85 if (!ractx->fdsp)

86 return AVERROR(ENOMEM);

88 return 0;

89 }

91 static void convolve(float *tgt, const float *src, int len, int n)

92 {

93 for (; n >= 0; n--)

94 tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);

96 }

98 static void decode(RA288Context *ractx, float gain, int cb_coef)

99 {

100 int i;

101 double sumsum;

102 float sum, buffer[5];

103 float *block = ractx->sp_hist + 70 + 36; // current block

104 float *gain_block = ractx->gain_hist + 28;

105

106 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));

107

108 /* block 46 of G.728 spec */

109 sum = 32.0;

110 for (i=0; i < 10; i++)

111 sum -= gain_block[9-i] * ractx->gain_lpc[i];

112

113 /* block 47 of G.728 spec */

114 sum = av_clipf(sum, 0, 60);

115

116 /* block 48 of G.728 spec */

117 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */

118 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));

119

120 for (i=0; i < 5; i++)

121 buffer[i] = codetable[cb_coef][i] * sumsum;

122

123 sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);

124

125 sum = FFMAX(sum, 5.0 / (1<<24));

126

127 /* shift and store */

128 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));

129

130 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);

131

132 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);

133 }

134

135 /**

136 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.

137 *

138 * @param order filter order

139 * @param n input length

140 * @param non_rec number of non-recursive samples

141 * @param out filter output

142 * @param hist pointer to the input history of the filter

143 * @param out pointer to the non-recursive part of the output

144 * @param out2 pointer to the recursive part of the output

145 * @param window pointer to the windowing function table

146 */

147 static void do_hybrid_window(RA288Context *ractx,

148 int order, int n, int non_rec, float *out,

149 float *hist, float *out2, const float *window)

150 {

151 int i;

152 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];

153 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];

154 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +

155 MAX_BACKWARD_FILTER_LEN +

156 MAX_BACKWARD_FILTER_NONREC, 16)]);

157

158 av_assert2(order>=0);

159

160 ractx->fdsp->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));

161

162 convolve(buffer1, work + order , n , order);

163 convolve(buffer2, work + order + n, non_rec, order);

164

165 for (i=0; i <= order; i++) {

166 out2[i] = out2[i] * 0.5625 + buffer1[i];

167 out [i] = out2[i] + buffer2[i];

168 }

169

170 /* Multiply by the white noise correcting factor (WNCF). */

171 *out *= 257.0 / 256.0;

172 }

173

174 /**

175 * Backward synthesis filter, find the LPC coefficients from past speech data.

176 */

177 static void backward_filter(RA288Context *ractx,

178 float *hist, float *rec, const float *window,

179 float *lpc, const float *tab,

180 int order, int n, int non_rec, int move_size)

181 {

182 float temp[MAX_BACKWARD_FILTER_ORDER+1];

183

184 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);

185

186 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))

187 ractx->fdsp->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));

188

189 memmove(hist, hist + n, move_size*sizeof(*hist));

190 }

191

192 static int ra288_decode_frame(AVCodecContext * avctx, void *data,

193 int *got_frame_ptr, AVPacket *avpkt)

194 {

195 AVFrame *frame = data;

196 const uint8_t *buf = avpkt->data;

197 int buf_size = avpkt->size;

198 float *out;

199 int i, ret;

200 RA288Context *ractx = avctx->priv_data;

201 GetBitContext gb;

202

203 if (buf_size < avctx->block_align) {

204 av_log(avctx, AV_LOG_ERROR,

205 "Error! Input buffer is too small [%d<%d]\n",

206 buf_size, avctx->block_align);

207 return AVERROR_INVALIDDATA;

208 }

209

210 /* get output buffer */

211 frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;

212 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

213 return ret;

214 out = (float *)frame->data[0];

215

216 init_get_bits8(&gb, buf, avctx->block_align);

217

218 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {

219 float gain = amptable[get_bits(&gb, 3)];

220 int cb_coef = get_bits(&gb, 6 + (i&1));

221

222 decode(ractx, gain, cb_coef);

223

224 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));

225 out += RA288_BLOCK_SIZE;

226

227 if ((i & 7) == 3) {

228 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,

229 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);

230

231 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,

232 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);

233 }

234 }

235

236 *got_frame_ptr = 1;

237

238 return avctx->block_align;

239 }

240

241 AVCodec ff_ra_288_decoder = {

242 .name = "real_288",

243 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),

244 .type = AVMEDIA_TYPE_AUDIO,

245 .id = AV_CODEC_ID_RA_288,

246 .priv_data_size = sizeof(RA288Context),

247 .init = ra288_decode_init,

248 .decode = ra288_decode_frame,

249 .close = ra288_decode_close,

250 .capabilities = CODEC_CAP_DR1,

251 };

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