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authorRoman Shaposhnik <roman@shaposhnik.org>2004-02-07 08:20:00 +0000
committerRoman Shaposhnik <roman@shaposhnik.org>2004-02-07 08:20:00 +0000
commite5966052ee7e07c30a078f31141e3e068847fc7a (patch)
tree5457e8baa39f57fa705e23bc037aafcc9dbb47aa
parent2d9aa1e3a24e8640e0527a4def4b5e93a80142a4 (diff)
downloadffmpeg-e5966052ee7e07c30a078f31141e3e068847fc7a.tar.gz
* Initial implementation of the G.726 ADPCM audio codec.
Originally committed as revision 2759 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/Makefile2
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/avcodec.h2
-rw-r--r--libavcodec/g726.c417
-rw-r--r--libavformat/wav.c1
5 files changed, 422 insertions, 1 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 3d4c321d44..8bfaf7efe5 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -19,7 +19,7 @@ OBJS= common.o utils.o mem.o allcodecs.o \
vp3.o asv1.o 4xm.o cabac.o ffv1.o ra144.o ra288.o vcr1.o cljr.o \
roqvideo.o dpcm.o interplayvideo.o xan.o rpza.o cinepak.o msrle.o \
msvideo1.o vqavideo.o idcinvideo.o adx.o rational.o faandct.o 8bps.o \
- smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o
+ smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o g726.o
ifeq ($(AMR_NB),yes)
ifeq ($(AMR_NB_FIXED),yes)
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 4abc462e90..5fe6dae62b 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -203,6 +203,7 @@ PCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm);
PCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
PCM_CODEC(CODEC_ID_ADPCM_ADX, adpcm_adx);
PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
+PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_g726);
#undef PCM_CODEC
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 342065a221..8f3b9dc640 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -117,6 +117,7 @@ enum CodecID {
CODEC_ID_ADPCM_XA,
CODEC_ID_ADPCM_ADX,
CODEC_ID_ADPCM_EA,
+ CODEC_ID_ADPCM_G726,
/* AMR */
CODEC_ID_AMR_NB,
@@ -1730,6 +1731,7 @@ PCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm);
PCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
PCM_CODEC(CODEC_ID_ADPCM_ADX, adpcm_adx);
PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
+PCM_CODEC(CODEC_ID_ADPCM_G726, adpcm_g726);
#undef PCM_CODEC
diff --git a/libavcodec/g726.c b/libavcodec/g726.c
new file mode 100644
index 0000000000..2166e3b5ef
--- /dev/null
+++ b/libavcodec/g726.c
@@ -0,0 +1,417 @@
+/*
+ * G.726 ADPCM audio codec
+ * Copyright (c) 2004 Roman Shaposhnik.
+ *
+ * This is a very straightforward rendition of the G.726
+ * Section 4 "Computational Details".
+ *
+ * This library 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 of the License, or (at your option) any later version.
+ *
+ * This library 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 this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+#include <limits.h>
+#include "avcodec.h"
+#include "common.h"
+
+/*
+ * G.726 Standard uses rather odd 11bit floating point arithmentic for
+ * numerous occasions. It's a mistery to me why they did it this way
+ * instead of simply using 32bit integer arithmetic.
+ */
+typedef struct Float11 {
+ int sign; /* 1bit sign */
+ int exp; /* 4bit exponent */
+ int mant; /* 6bit mantissa */
+} Float11;
+
+static inline Float11* i2f(int16_t i, Float11* f)
+{
+ f->sign = (i < 0);
+ if (f->sign)
+ i = -i;
+ f->exp = av_log2_16bit(i) + !!i;
+ f->mant = i? (i<<6) >> f->exp :
+ 1<<5;
+ return f;
+}
+
+static inline int16_t mult(Float11* f1, Float11* f2)
+{
+ int res, exp;
+
+ exp = f1->exp + f2->exp;
+ res = (((f1->mant * f2->mant) + 0x30) >> 4) << 7;
+ res = exp > 26 ? res << (exp - 26) : res >> (26 - exp);
+ return (f1->sign ^ f2->sign) ? -res : res;
+}
+
+static inline int clamp(int value, int min, int max)
+{
+ if (value < min)
+ return min;
+ else if (value > max)
+ return max;
+ else
+ return value;
+}
+
+static inline int sgn(int value)
+{
+ return (value < 0) ? -1 : 1;
+}
+
+typedef struct G726Tables {
+ int bits; /* bits per sample */
+ int* quant; /* quantization table */
+ int* iquant; /* inverse quantization table */
+ int* W; /* special table #1 ;-) */
+ int* F; /* special table #2 */
+} G726Tables;
+
+typedef struct G726Context {
+ G726Tables* tbls; /* static tables needed for computation */
+
+ Float11 sr[2]; /* prev. reconstructed samples */
+ Float11 dq[6]; /* prev. difference */
+ int a[2]; /* second order predictor coeffs */
+ int b[6]; /* sixth order predictor coeffs */
+ int pk[2]; /* signs of prev. 2 sez + dq */
+
+ int ap; /* scale factor control */
+ int yu; /* fast scale factor */
+ int yl; /* slow scale factor */
+ int dms; /* short average magnitude of F[i] */
+ int dml; /* long average magnitude of F[i] */
+ int td; /* tone detect */
+
+ int se; /* estimated signal for the next iteration */
+ int sez; /* estimated second order prediction */
+ int y; /* quantizer scaling factor for the next iteration */
+} G726Context;
+
+static int quant_tbl16[] = /* 16kbit/s 2bits per sample */
+ { 260, INT_MAX };
+static int iquant_tbl16[] =
+ { 116, 365, 365, 116 };
+static int W_tbl16[] =
+ { -22, 439, 439, -22 };
+static int F_tbl16[] =
+ { 0, 7, 7, 0 };
+
+static int quant_tbl24[] = /* 24kbit/s 3bits per sample */
+ { 7, 217, 330, INT_MAX };
+static int iquant_tbl24[] =
+ { INT_MIN, 135, 273, 373, 373, 273, 135, INT_MIN };
+static int W_tbl24[] =
+ { -4, 30, 137, 582, 582, 137, 30, -4 };
+static int F_tbl24[] =
+ { 0, 1, 2, 7, 7, 2, 1, 0 };
+
+static int quant_tbl32[] = /* 32kbit/s 4bits per sample */
+ { -125, 79, 177, 245, 299, 348, 399, INT_MAX };
+static int iquant_tbl32[] =
+ { INT_MIN, 4, 135, 213, 273, 323, 373, 425,
+ 425, 373, 323, 273, 213, 135, 4, INT_MIN };
+static int W_tbl32[] =
+ { -12, 18, 41, 64, 112, 198, 355, 1122,
+ 1122, 355, 198, 112, 64, 41, 18, -12};
+static int F_tbl32[] =
+ { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
+
+static int quant_tbl40[] = /* 40kbit/s 5bits per sample */
+ { -122, -16, 67, 138, 197, 249, 297, 338,
+ 377, 412, 444, 474, 501, 527, 552, INT_MAX };
+static int iquant_tbl40[] =
+ { INT_MIN, -66, 28, 104, 169, 224, 274, 318,
+ 358, 395, 429, 459, 488, 514, 539, 566,
+ 566, 539, 514, 488, 459, 429, 395, 358,
+ 318, 274, 224, 169, 104, 28, -66, INT_MIN };
+static int W_tbl40[] =
+ { 14, 14, 24, 39, 40, 41, 58, 100,
+ 141, 179, 219, 280, 358, 440, 529, 696,
+ 696, 529, 440, 358, 280, 219, 179, 141,
+ 100, 58, 41, 40, 39, 24, 14, 14 };
+static int F_tbl40[] =
+ { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
+ 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
+
+static G726Tables G726Tables_pool[] =
+ {{ 2, quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
+ { 3, quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
+ { 4, quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
+ { 5, quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
+
+
+/*
+ * Para 4.2.2 page 18: Adaptive quantizer.
+ */
+static inline uint8_t quant(G726Context* c, int d)
+{
+ int sign, exp, i, dln;
+
+ sign = i = 0;
+ if (d < 0) {
+ sign = 1;
+ d = -d;
+ }
+ exp = av_log2_16bit(d);
+ dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
+
+ while (c->tbls->quant[i] < INT_MAX && c->tbls->quant[i] < dln)
+ ++i;
+
+ if (sign)
+ i = ~i;
+ if (c->tbls->bits != 2 && i == 0) /* I'm not sure this is a good idea */
+ i = 0xff;
+
+ return i;
+}
+
+/*
+ * Para 4.2.3 page 22: Inverse adaptive quantizer.
+ */
+static inline int16_t inverse_quant(G726Context* c, int i)
+{
+ int dql, dex, dqt;
+
+ dql = c->tbls->iquant[i] + (c->y >> 2);
+ dex = (dql>>7) & 0xf; /* 4bit exponent */
+ dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
+ return (dql < 0) ? 0 : ((dqt<<7) >> (14-dex));
+}
+
+static inline int16_t g726_iterate(G726Context* c, int16_t I)
+{
+ int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
+ Float11 f;
+
+ dq = inverse_quant(c, I);
+ if (I >> (c->tbls->bits - 1)) /* get the sign */
+ dq = -dq;
+ re_signal = c->se + dq;
+
+ /* Transition detect */
+ ylint = (c->yl >> 15);
+ ylfrac = (c->yl >> 10) & 0x1f;
+ thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
+ if (c->td == 1 && abs(dq) > ((thr2+(thr2>>1))>>1))
+ tr = 1;
+ else
+ tr = 0;
+
+ /* Update second order predictor coefficient A2 and A1 */
+ pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
+ dq0 = dq ? sgn(dq) : 0;
+ if (tr) {
+ c->a[0] = 0;
+ c->a[1] = 0;
+ for (i=0; i<6; i++)
+ c->b[i] = 0;
+ } else {
+ /* This is a bit crazy, but it really is +255 not +256 */
+ fa1 = clamp((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
+
+ c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
+ c->a[1] = clamp(c->a[1], -12288, 12288);
+ c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
+ c->a[0] = clamp(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
+
+ for (i=0; i<6; i++)
+ c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
+ }
+
+ /* Update Dq and Sr and Pk */
+ c->pk[1] = c->pk[0];
+ c->pk[0] = pk0 ? pk0 : 1;
+ c->sr[1] = c->sr[0];
+ i2f(re_signal, &c->sr[0]);
+ for (i=5; i>0; i--)
+ c->dq[i] = c->dq[i-1];
+ i2f(dq, &c->dq[0]);
+ c->dq[0].sign = I >> (c->tbls->bits - 1); /* Isn't it crazy ?!?! */
+
+ /* Update tone detect [I'm not sure 'tr == 0' is really needed] */
+ c->td = (tr == 0 && c->a[1] < -11776);
+
+ /* Update Ap */
+ c->dms += ((c->tbls->F[I]<<9) - c->dms) >> 5;
+ c->dml += ((c->tbls->F[I]<<11) - c->dml) >> 7;
+ if (tr)
+ c->ap = 256;
+ else if (c->y > 1535 && !c->td && (abs((c->dms << 2) - c->dml) < (c->dml >> 3)))
+ c->ap += (-c->ap) >> 4;
+ else
+ c->ap += (0x200 - c->ap) >> 4;
+
+ /* Update Yu and Yl */
+ c->yu = clamp(c->y + (((c->tbls->W[I] << 5) - c->y) >> 5), 544, 5120);
+ c->yl += c->yu + ((-c->yl)>>6);
+
+ /* Next iteration for Y */
+ al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
+ c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
+
+ /* Next iteration for SE and SEZ */
+ c->se = 0;
+ for (i=0; i<6; i++)
+ c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
+ c->sez = c->se >> 1;
+ for (i=0; i<2; i++)
+ c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
+ c->se >>= 1;
+
+ return clamp(re_signal << 2, -0xffff, 0xffff);
+}
+
+static int g726_reset(G726Context* c, int bit_rate)
+{
+ int i;
+
+ c->tbls = &G726Tables_pool[bit_rate/8000 - 2];
+ for (i=0; i<2; i++) {
+ i2f(0, &c->sr[i]);
+ c->a[i] = 0;
+ c->pk[i] = 1;
+ }
+ for (i=0; i<6; i++) {
+ i2f(0, &c->dq[i]);
+ c->b[i] = 0;
+ }
+ c->ap = 0;
+ c->dms = 0;
+ c->dml = 0;
+ c->yu = 544;
+ c->yl = 34816;
+ c->td = 0;
+
+ c->se = 0;
+ c->sez = 0;
+ c->y = 544;
+
+ return 0;
+}
+
+static int16_t g726_decode(G726Context* c, int16_t i)
+{
+ return g726_iterate(c, i);
+}
+
+static int16_t g726_encode(G726Context* c, int16_t sig)
+{
+ uint8_t i;
+
+ i = quant(c, sig/4 - c->se) & ((1<<c->tbls->bits) - 1);
+ g726_iterate(c, i);
+ return i;
+}
+
+/* Interfacing to the libavcodec */
+
+typedef struct AVG726Context {
+ G726Context c;
+ int bits_left;
+ int bit_buffer;
+ int code_size;
+} AVG726Context;
+
+static int g726_init(AVCodecContext * avctx)
+{
+ AVG726Context* c = (AVG726Context*)avctx->priv_data;
+
+ if (avctx->sample_rate != 8000 || avctx->channels != 1 ||
+ (avctx->bit_rate != 16000 && avctx->bit_rate != 24000 &&
+ avctx->bit_rate != 32000 && avctx->bit_rate != 40000)) {
+ av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n");
+ return -1;
+ }
+ g726_reset(&c->c, avctx->bit_rate);
+ c->code_size = c->c.tbls->bits;
+ c->bit_buffer = 0;
+ c->bits_left = 0;
+
+ return 0;
+}
+
+static int g726_encode_frame(AVCodecContext *avctx,
+ uint8_t *dst, int buf_size, void *data)
+{
+ AVG726Context *c = avctx->priv_data;
+ short *samples = data;
+ PutBitContext pb;
+
+ init_put_bits(&pb, dst, 1024*1024);
+
+ for (; buf_size; buf_size--)
+ put_bits(&pb, c->code_size, g726_encode(&c->c, *samples++));
+
+ flush_put_bits(&pb);
+
+ return put_bits_count(&pb)>>3;
+}
+
+static int g726_decode_frame(AVCodecContext *avctx,
+ void *data, int *data_size,
+ uint8_t *buf, int buf_size)
+{
+ AVG726Context *c = avctx->priv_data;
+ short *samples = data;
+ uint8_t code;
+ uint8_t mask;
+ GetBitContext gb;
+
+ if (!buf_size)
+ goto out;
+
+ mask = (1<<c->code_size) - 1;
+ init_get_bits(&gb, buf, buf_size * 8);
+ if (c->bits_left) {
+ int s = c->code_size - c->bits_left;;
+ code = (c->bit_buffer << s) | get_bits(&gb, s);
+ *samples++ = g726_decode(&c->c, code & mask);
+ }
+
+ while (get_bits_count(&gb) + c->code_size <= buf_size*8)
+ *samples++ = g726_decode(&c->c, get_bits(&gb, c->code_size) & mask);
+
+ c->bits_left = buf_size*8 - get_bits_count(&gb);
+ c->bit_buffer = get_bits(&gb, c->bits_left);
+
+out:
+ *data_size = (uint8_t*)samples - (uint8_t*)data;
+ return buf_size;
+}
+
+#ifdef CONFIG_ENCODERS
+AVCodec adpcm_g726_encoder = {
+ "g726",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_ADPCM_G726,
+ sizeof(AVG726Context),
+ g726_init,
+ g726_encode_frame,
+ NULL,
+ NULL,
+};
+#endif //CONFIG_ENCODERS
+
+AVCodec adpcm_g726_decoder = {
+ "g726",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_ADPCM_G726,
+ sizeof(AVG726Context),
+ g726_init,
+ NULL,
+ NULL,
+ g726_decode_frame,
+};
diff --git a/libavformat/wav.c b/libavformat/wav.c
index 22e605fb48..787c7d542f 100644
--- a/libavformat/wav.c
+++ b/libavformat/wav.c
@@ -29,6 +29,7 @@ const CodecTag codec_wav_tags[] = {
{ CODEC_ID_PCM_MULAW, 0x07 },
{ CODEC_ID_ADPCM_MS, 0x02 },
{ CODEC_ID_ADPCM_IMA_WAV, 0x11 },
+ { CODEC_ID_ADPCM_G726, 0x45 },
{ CODEC_ID_ADPCM_IMA_DK4, 0x61 }, /* rogue format number */
{ CODEC_ID_ADPCM_IMA_DK3, 0x62 }, /* rogue format number */
{ CODEC_ID_WMAV1, 0x160 },