/*
* G.726 ADPCM audio codec
- * Copyright (c) 2004 Roman Shaposhnik.
+ * 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
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <limits.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+
#include "avcodec.h"
-#include "common.h"
#include "bitstream.h"
+#include "internal.h"
+#include "put_bits.h"
/**
- * G.726 11bit float.
- * 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.
+ * G.726 11-bit float.
+ * G.726 Standard uses rather odd 11-bit floating point arithmetic for
+ * numerous occasions. It's a mystery to me why they did it this way
+ * instead of simply using 32-bit integer arithmetic.
*/
typedef struct Float11 {
- int sign; /**< 1bit sign */
- int exp; /**< 4bit exponent */
- int mant; /**< 6bit mantissa */
+ uint8_t sign; /**< 1 bit sign */
+ uint8_t exp; /**< 4 bits exponent */
+ uint8_t mant; /**< 6 bits mantissa */
} Float11;
-static inline Float11* i2f(int16_t i, Float11* f)
+static inline Float11* i2f(int 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;
+ 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);
+ res = (((f1->mant * f2->mant) + 0x30) >> 4);
+ res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
return (f1->sign ^ f2->sign) ? -res : res;
}
static inline int sgn(int value)
{
- return (value < 0) ? -1 : 1;
+ 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 */
+ const int* quant; /**< quantization table */
+ const int16_t* iquant; /**< inverse quantization table */
+ const int16_t* W; /**< special table #1 ;-) */
+ const uint8_t* 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 */
+ AVClass *class;
+ 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 */
+ int code_size;
} G726Context;
-static int quant_tbl16[] = /**< 16kbit/s 2bits per sample */
+static const int quant_tbl16[] = /**< 16kbit/s 2 bits per sample */
{ 260, INT_MAX };
-static int iquant_tbl16[] =
+static const int16_t iquant_tbl16[] =
{ 116, 365, 365, 116 };
-static int W_tbl16[] =
+static const int16_t W_tbl16[] =
{ -22, 439, 439, -22 };
-static int F_tbl16[] =
+static const uint8_t F_tbl16[] =
{ 0, 7, 7, 0 };
-static int quant_tbl24[] = /**< 24kbit/s 3bits per sample */
+static const int quant_tbl24[] = /**< 24kbit/s 3 bits 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[] =
+static const int16_t iquant_tbl24[] =
+ { INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
+static const int16_t W_tbl24[] =
{ -4, 30, 137, 582, 582, 137, 30, -4 };
-static int F_tbl24[] =
+static const uint8_t F_tbl24[] =
{ 0, 1, 2, 7, 7, 2, 1, 0 };
-static int quant_tbl32[] = /**< 32kbit/s 4bits per sample */
+static const int quant_tbl32[] = /**< 32kbit/s 4 bits 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[] =
+static const int16_t iquant_tbl32[] =
+ { INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
+ 425, 373, 323, 273, 213, 135, 4, INT16_MIN };
+static const int16_t W_tbl32[] =
{ -12, 18, 41, 64, 112, 198, 355, 1122,
1122, 355, 198, 112, 64, 41, 18, -12};
-static int F_tbl32[] =
+static const uint8_t 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 */
+static const int quant_tbl40[] = /**< 40kbit/s 5 bits 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,
+static const int16_t iquant_tbl40[] =
+ { INT16_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[] =
+ 318, 274, 224, 169, 104, 28, -66, INT16_MIN };
+static const int16_t 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[] =
+static const uint8_t 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 }};
+static const G726Tables G726Tables_pool[] =
+ {{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
+ { quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
+ { quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
+ { quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
/**
- * Para 4.2.2 page 18: Adaptive quantizer.
+ * Paragraph 4.2.2 page 18: Adaptive quantizer.
*/
static inline uint8_t quant(G726Context* c, int d)
{
- int sign, exp, i, dln;
+ 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);
+ 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)
+ 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;
+ if (sign)
+ i = ~i;
+ if (c->code_size != 2 && i == 0) /* I'm not sure this is a good idea */
+ i = 0xff;
- return i;
+ return i;
}
/**
- * Para 4.2.3 page 22: Inverse adaptive quantizer.
+ * Paragraph 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 */
+ dql = c->tbls.iquant[i] + (c->y >> 2);
+ dex = (dql>>7) & 0xf; /* 4-bit exponent */
dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
- return (dql < 0) ? 0 : ((dqt<<7) >> (14-dex));
+ return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
}
-static inline int16_t g726_iterate(G726Context* c, int16_t I)
+static int16_t g726_decode(G726Context* c, int I)
{
int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
Float11 f;
+ int I_sig= I >> (c->code_size - 1);
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;
+ tr= (c->td == 1 && dq > ((3*thr2)>>2));
+
+ if (I_sig) /* get the sign */
+ dq = -dq;
+ re_signal = c->se + dq;
/* Update second order predictor coefficient A2 and A1 */
pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
c->a[0] = 0;
c->a[1] = 0;
for (i=0; i<6; i++)
- c->b[i] = 0;
+ c->b[i] = 0;
} else {
/* This is a bit crazy, but it really is +255 not +256 */
- fa1 = clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
+ fa1 = av_clip_intp2((-c->a[0]*c->pk[0]*pk0)>>5, 8);
c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
- c->a[1] = clip(c->a[1], -12288, 12288);
+ c->a[1] = av_clip(c->a[1], -12288, 12288);
c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
- c->a[0] = clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
+ c->a[0] = av_clip(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);
+ c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
}
/* Update Dq and Sr and Pk */
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];
+ 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 ?!?! */
+ c->dq[0].sign = I_sig; /* Isn't it crazy ?!?! */
- /* Update tone detect [I'm not sure 'tr == 0' is really needed] */
- c->td = (tr == 0 && c->a[1] < -11776);
+ c->td = 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;
+ c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5);
+ c->dml += (c->tbls.F[I]<<4) + ((- 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;
+ c->ap = 256;
+ else {
+ c->ap += (-c->ap) >> 4;
+ if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3))
+ c->ap += 0x20;
+ }
/* Update Yu and Yl */
- c->yu = clip(c->y + (((c->tbls->W[I] << 5) - c->y) >> 5), 544, 5120);
+ c->yu = av_clip(c->y + c->tbls.W[I] + ((-c->y)>>5), 544, 5120);
c->yl += c->yu + ((-c->yl)>>6);
/* Next iteration for Y */
/* 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->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 += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
c->se >>= 1;
- return clip(re_signal << 2, -0xffff, 0xffff);
+ return av_clip(re_signal << 2, -0xffff, 0xffff);
}
-static int g726_reset(G726Context* c, int bit_rate)
+static av_cold int g726_reset(G726Context *c)
{
int i;
- c->tbls = &G726Tables_pool[bit_rate/8000 - 2];
+ c->tbls = G726Tables_pool[c->code_size - 2];
for (i=0; i<2; i++) {
- i2f(0, &c->sr[i]);
- c->a[i] = 0;
- c->pk[i] = 1;
+ c->sr[i].mant = 1<<5;
+ c->pk[i] = 1;
}
for (i=0; i<6; i++) {
- i2f(0, &c->dq[i]);
- c->b[i] = 0;
+ c->dq[i].mant = 1<<5;
}
- 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);
-}
-
+#if CONFIG_ADPCM_G726_ENCODER
static int16_t g726_encode(G726Context* c, int16_t sig)
{
- uint8_t i;
+ uint8_t i;
- i = quant(c, sig/4 - c->se) & ((1<<c->tbls->bits) - 1);
- g726_iterate(c, i);
- return i;
+ i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
+ g726_decode(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)
+static av_cold int g726_encode_init(AVCodecContext *avctx)
{
- AVG726Context* c = (AVG726Context*)avctx->priv_data;
-
- if (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;
+ G726Context* c = avctx->priv_data;
+
+ if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
+ avctx->sample_rate != 8000) {
+ av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
+ "allowed when the compliance level is higher than unofficial. "
+ "Resample or reduce the compliance level.\n");
+ return AVERROR(EINVAL);
+ }
+ if (avctx->sample_rate <= 0) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid sample rate %d\n",
+ avctx->sample_rate);
+ return AVERROR(EINVAL);
}
- if (avctx->sample_rate != 8000 && avctx->strict_std_compliance>FF_COMPLIANCE_INOFFICIAL) {
- av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n");
- return -1;
+
+ if(avctx->channels != 1){
+ av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
+ return AVERROR(EINVAL);
}
- g726_reset(&c->c, avctx->bit_rate);
- c->code_size = c->c.tbls->bits;
- c->bit_buffer = 0;
- c->bits_left = 0;
- avctx->coded_frame = avcodec_alloc_frame();
- if (!avctx->coded_frame)
- return -ENOMEM;
- avctx->coded_frame->key_frame = 1;
+ if (avctx->bit_rate)
+ c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
+
+ c->code_size = av_clip(c->code_size, 2, 5);
+ avctx->bit_rate = c->code_size * avctx->sample_rate;
+ avctx->bits_per_coded_sample = c->code_size;
+
+ g726_reset(c);
+
+ /* select a frame size that will end on a byte boundary and have a size of
+ approximately 1024 bytes */
+ avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
return 0;
}
-static int g726_close(AVCodecContext *avctx)
+static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
- av_freep(&avctx->coded_frame);
+ G726Context *c = avctx->priv_data;
+ const int16_t *samples = (const int16_t *)frame->data[0];
+ PutBitContext pb;
+ int i, ret, out_size;
+
+ out_size = (frame->nb_samples * c->code_size + 7) / 8;
+ if ((ret = ff_alloc_packet(avpkt, out_size))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+ init_put_bits(&pb, avpkt->data, avpkt->size);
+
+ for (i = 0; i < frame->nb_samples; i++)
+ put_bits(&pb, c->code_size, g726_encode(c, *samples++));
+
+ flush_put_bits(&pb);
+
+ avpkt->size = out_size;
+ *got_packet_ptr = 1;
return 0;
}
-static int g726_encode_frame(AVCodecContext *avctx,
- uint8_t *dst, int buf_size, void *data)
+#define OFFSET(x) offsetof(G726Context, x)
+#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
+static const AVOption options[] = {
+ { "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { .i64 = 4 }, 2, 5, AE },
+ { NULL },
+};
+
+static const AVClass class = {
+ .class_name = "g726",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+static const AVCodecDefault defaults[] = {
+ { "b", "0" },
+ { NULL },
+};
+
+AVCodec ff_adpcm_g726_encoder = {
+ .name = "g726",
+ .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_ADPCM_G726,
+ .priv_data_size = sizeof(G726Context),
+ .init = g726_encode_init,
+ .encode2 = g726_encode_frame,
+ .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .priv_class = &class,
+ .defaults = defaults,
+};
+#endif
+
+#if CONFIG_ADPCM_G726_DECODER
+static av_cold int g726_decode_init(AVCodecContext *avctx)
{
- AVG726Context *c = avctx->priv_data;
- short *samples = data;
- PutBitContext pb;
+ G726Context* c = avctx->priv_data;
- init_put_bits(&pb, dst, 1024*1024);
+ avctx->channels = 1;
+ avctx->channel_layout = AV_CH_LAYOUT_MONO;
- for (; buf_size; buf_size--)
- put_bits(&pb, c->code_size, g726_encode(&c->c, *samples++));
+ c->code_size = avctx->bits_per_coded_sample;
+ if (c->code_size < 2 || c->code_size > 5) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
+ return AVERROR(EINVAL);
+ }
+ g726_reset(c);
- flush_put_bits(&pb);
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- return put_bits_count(&pb)>>3;
+ return 0;
}
-static int g726_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- uint8_t *buf, int buf_size)
+static int g726_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
- 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);
+ AVFrame *frame = data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ G726Context *c = avctx->priv_data;
+ int16_t *samples;
+ BitstreamContext bc;
+ int out_samples, ret;
+
+ out_samples = buf_size * 8 / c->code_size;
+
+ /* get output buffer */
+ frame->nb_samples = out_samples;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
+ samples = (int16_t *)frame->data[0];
+
+ bitstream_init8(&bc, buf, buf_size);
- while (get_bits_count(&gb) + c->code_size <= buf_size*8)
- *samples++ = g726_decode(&c->c, get_bits(&gb, c->code_size) & mask);
+ while (out_samples--)
+ *samples++ = g726_decode(c, bitstream_read(&bc, c->code_size));
- c->bits_left = buf_size*8 - get_bits_count(&gb);
- c->bit_buffer = get_bits(&gb, c->bits_left);
+ if (bitstream_bits_left(&bc) > 0)
+ av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
+
+ *got_frame_ptr = 1;
-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,
- g726_close,
- NULL,
-};
-#endif //CONFIG_ENCODERS
-
-AVCodec adpcm_g726_decoder = {
- "g726",
- CODEC_TYPE_AUDIO,
- CODEC_ID_ADPCM_G726,
- sizeof(AVG726Context),
- g726_init,
- NULL,
- g726_close,
- g726_decode_frame,
+static void g726_decode_flush(AVCodecContext *avctx)
+{
+ G726Context *c = avctx->priv_data;
+ g726_reset(c);
+}
+
+AVCodec ff_adpcm_g726_decoder = {
+ .name = "g726",
+ .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_ADPCM_G726,
+ .priv_data_size = sizeof(G726Context),
+ .init = g726_decode_init,
+ .decode = g726_decode_frame,
+ .flush = g726_decode_flush,
+ .capabilities = AV_CODEC_CAP_DR1,
};
+#endif