[FFmpeg-devel] [PATCH]Split G.726 encoder and decoder

Carl Eugen Hoyos cehoyos at ag.or.at
Thu Jun 6 12:59:12 CEST 2013


Hi!

Attached patch splits the G.726 encoder and decoder from the common code.

Please comment, Carl Eugen
-------------- next part --------------
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 1aeea28..2c8c50b 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -583,8 +583,8 @@ OBJS-$(CONFIG_ADPCM_EA_R3_DECODER)        += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_G722_DECODER)         += g722.o g722dec.o
 OBJS-$(CONFIG_ADPCM_G722_ENCODER)         += g722.o g722enc.o
-OBJS-$(CONFIG_ADPCM_G726_DECODER)         += g726.o
-OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o
+OBJS-$(CONFIG_ADPCM_G726_DECODER)         += g726.o g726dec.o
+OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o g726enc.o
 OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER)      += adpcm.o adpcm_data.o
diff --git a/libavcodec/g726.c b/libavcodec/g726.c
index 58d0468..4b2a1eb 100644
--- a/libavcodec/g726.c
+++ b/libavcodec/g726.c
@@ -22,6 +22,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include <limits.h>
+#include "g726.h"
 
 #include "libavutil/avassert.h"
 #include "libavutil/channel_layout.h"
@@ -31,18 +32,6 @@
 #include "get_bits.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 mystery to me why they did it this way
- * instead of simply using 32bit integer arithmetic.
- */
-typedef struct Float11 {
-    uint8_t sign;   /**< 1bit sign */
-    uint8_t exp;    /**< 4bit exponent */
-    uint8_t mant;   /**< 6bit mantissa */
-} Float11;
-
 static inline Float11* i2f(int i, Float11* f)
 {
     f->sign = (i < 0);
@@ -68,36 +57,6 @@ static inline int sgn(int value)
     return (value < 0) ? -1 : 1;
 }
 
-typedef struct G726Tables {
-    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 {
-    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 const int quant_tbl16[] =                  /**< 16kbit/s 2bits per sample */
            { 260, INT_MAX };
 static const int16_t iquant_tbl16[] =
@@ -152,32 +111,6 @@ static const G726Tables G726Tables_pool[] =
 
 
 /**
- * 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->code_size != 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)
@@ -190,7 +123,7 @@ static inline int16_t inverse_quant(G726Context* c, int i)
     return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
 }
 
-static int16_t g726_decode(G726Context* c, int I)
+int16_t ff_g726_decode(G726Context* c, int I)
 {
     int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
     Float11 f;
@@ -272,7 +205,7 @@ static int16_t g726_decode(G726Context* c, int I)
     return av_clip(re_signal << 2, -0xffff, 0xffff);
 }
 
-static av_cold int g726_reset(G726Context *c)
+int ff_g726_reset(G726Context *c)
 {
     int i;
 
@@ -291,178 +224,3 @@ static av_cold int g726_reset(G726Context *c)
 
     return 0;
 }
-
-#if CONFIG_ADPCM_G726_ENCODER
-static int16_t g726_encode(G726Context* c, int16_t sig)
-{
-    uint8_t i;
-
-    i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
-    g726_decode(c, i);
-    return i;
-}
-
-/* Interfacing to the libavcodec */
-
-static av_cold int g726_encode_init(AVCodecContext *avctx)
-{
-    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);
-    }
-    av_assert0(avctx->sample_rate > 0);
-
-    if(avctx->channels != 1){
-        av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
-        return AVERROR(EINVAL);
-    }
-
-    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_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                             const AVFrame *frame, int *got_packet_ptr)
-{
-    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_packet2(avctx, avpkt, out_size)) < 0)
-        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;
-}
-
-#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",
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_ADPCM_G726,
-    .priv_data_size = sizeof(G726Context),
-    .init           = g726_encode_init,
-    .encode2        = g726_encode_frame,
-    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
-                                                     AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
-    .priv_class     = &class,
-    .defaults       = defaults,
-};
-#endif
-
-#if CONFIG_ADPCM_G726_DECODER
-static av_cold int g726_decode_init(AVCodecContext *avctx)
-{
-    G726Context* c = avctx->priv_data;
-
-    avctx->channels       = 1;
-    avctx->channel_layout = AV_CH_LAYOUT_MONO;
-
-    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);
-
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-
-    return 0;
-}
-
-static int g726_decode_frame(AVCodecContext *avctx, void *data,
-                             int *got_frame_ptr, AVPacket *avpkt)
-{
-    AVFrame *frame     = data;
-    const uint8_t *buf = avpkt->data;
-    int buf_size = avpkt->size;
-    G726Context *c = avctx->priv_data;
-    int16_t *samples;
-    GetBitContext gb;
-    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)
-        return ret;
-    samples = (int16_t *)frame->data[0];
-
-    init_get_bits(&gb, buf, buf_size * 8);
-
-    while (out_samples--)
-        *samples++ = g726_decode(c, get_bits(&gb, c->code_size));
-
-    if (get_bits_left(&gb) > 0)
-        av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
-
-    *got_frame_ptr = 1;
-
-    return buf_size;
-}
-
-static void g726_decode_flush(AVCodecContext *avctx)
-{
-    G726Context *c = avctx->priv_data;
-    g726_reset(c);
-}
-
-AVCodec ff_adpcm_g726_decoder = {
-    .name           = "g726",
-    .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   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
-};
-#endif
diff --git a/libavcodec/g726.h b/libavcodec/g726.h
new file mode 100644
index 0000000..1292faf
--- /dev/null
+++ b/libavcodec/g726.h
@@ -0,0 +1,77 @@
+/*
+ * 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 file is part of FFmpeg.
+ *
+ * FFmpeg 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 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#include <limits.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.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 mystery to me why they did it this way
+ * instead of simply using 32bit integer arithmetic.
+ */
+typedef struct Float11 {
+    uint8_t sign;   /**< 1bit sign */
+    uint8_t exp;    /**< 4bit exponent */
+    uint8_t mant;   /**< 6bit mantissa */
+} Float11;
+
+typedef struct G726Tables {
+    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 {
+    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;
+
+int16_t ff_g726_decode(G726Context* c, int I);
+int ff_g726_reset(G726Context *c);
diff --git a/libavcodec/g726dec.c b/libavcodec/g726dec.c
new file mode 100644
index 0000000..dd6e2b1
--- /dev/null
+++ b/libavcodec/g726dec.c
@@ -0,0 +1,102 @@
+/*
+ * 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 file is part of FFmpeg.
+ *
+ * FFmpeg 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 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#include <limits.h>
+#include "g726.h"
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "put_bits.h"
+
+static av_cold int g726_decode_init(AVCodecContext *avctx)
+{
+    G726Context* c = avctx->priv_data;
+
+    avctx->channels       = 1;
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
+
+    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);
+    }
+    ff_g726_reset(c);
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    return 0;
+}
+
+static int g726_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    AVFrame *frame     = data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    G726Context *c = avctx->priv_data;
+    int16_t *samples;
+    GetBitContext gb;
+    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)
+        return ret;
+    samples = (int16_t *)frame->data[0];
+
+    init_get_bits(&gb, buf, buf_size * 8);
+
+    while (out_samples--)
+        *samples++ = ff_g726_decode(c, get_bits(&gb, c->code_size));
+
+    if (get_bits_left(&gb) > 0)
+        av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
+
+    *got_frame_ptr = 1;
+
+    return buf_size;
+}
+
+static void g726_decode_flush(AVCodecContext *avctx)
+{
+    G726Context *c = avctx->priv_data;
+    ff_g726_reset(c);
+}
+
+AVCodec ff_adpcm_g726_decoder = {
+    .name           = "g726",
+    .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   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+};
diff --git a/libavcodec/g726enc.c b/libavcodec/g726enc.c
new file mode 100644
index 0000000..1925f3c
--- /dev/null
+++ b/libavcodec/g726enc.c
@@ -0,0 +1,161 @@
+/*
+ * 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 file is part of FFmpeg.
+ *
+ * FFmpeg 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 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#include <limits.h>
+#include "g726.h"
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "put_bits.h"
+
+/**
+ * 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->code_size != 2 && i == 0) /* I'm not sure this is a good idea */
+        i = 0xff;
+
+    return i;
+}
+
+static int16_t g726_encode(G726Context* c, int16_t sig)
+{
+    uint8_t i;
+
+    i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
+    ff_g726_decode(c, i);
+    return i;
+}
+
+/* Interfacing to the libavcodec */
+
+static av_cold int g726_encode_init(AVCodecContext *avctx)
+{
+    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);
+    }
+    av_assert0(avctx->sample_rate > 0);
+
+    if(avctx->channels != 1){
+        av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
+        return AVERROR(EINVAL);
+    }
+
+    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;
+
+    ff_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_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                             const AVFrame *frame, int *got_packet_ptr)
+{
+    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_packet2(avctx, avpkt, out_size)) < 0)
+        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;
+}
+
+#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",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ADPCM_G726,
+    .priv_data_size = sizeof(G726Context),
+    .init           = g726_encode_init,
+    .encode2        = g726_encode_frame,
+    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME,
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+                                                     AV_SAMPLE_FMT_NONE },
+    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+    .priv_class     = &class,
+    .defaults       = defaults,
+};


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