* This is a very straightforward rendition of the G.726
* Section 4 "Computational Details".
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg 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 FFmpeg; 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 "get_bits.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 {
- uint8_t sign; /**< 1bit sign */
- uint8_t exp; /**< 4bit exponent */
- uint8_t 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(int i, Float11* f)
} G726Tables;
typedef struct G726Context {
+ AVClass *class;
G726Tables tbls; /**< static tables needed for computation */
Float11 sr[2]; /**< prev. reconstructed samples */
int code_size;
} G726Context;
-static const int quant_tbl16[] = /**< 16kbit/s 2bits per sample */
+static const int quant_tbl16[] = /**< 16kbit/s 2 bits per sample */
{ 260, INT_MAX };
static const int16_t iquant_tbl16[] =
{ 116, 365, 365, 116 };
static const uint8_t F_tbl16[] =
{ 0, 7, 7, 0 };
-static const 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 const int16_t iquant_tbl24[] =
{ INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
static const uint8_t F_tbl24[] =
{ 0, 1, 2, 7, 7, 2, 1, 0 };
-static const 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 const int16_t iquant_tbl32[] =
{ INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
static const uint8_t F_tbl32[] =
{ 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
-static const 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 const int16_t iquant_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)
{
}
/**
- * 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 */
+ dex = (dql>>7) & 0xf; /* 4-bit exponent */
dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
}
c->b[i] = 0;
} else {
/* This is a bit crazy, but it really is +255 not +256 */
- fa1 = av_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] = av_clip(c->a[1], -12288, 12288);
return av_clip(re_signal << 2, -0xffff, 0xffff);
}
-static av_cold int g726_reset(G726Context* c, int index)
+static av_cold int g726_reset(G726Context *c)
{
int i;
- c->tbls = G726Tables_pool[index];
+ c->tbls = G726Tables_pool[c->code_size - 2];
for (i=0; i<2; i++) {
c->sr[i].mant = 1<<5;
c->pk[i] = 1;
g726_decode(c, i);
return i;
}
-#endif
/* Interfacing to the libavcodec */
-static av_cold int g726_init(AVCodecContext * avctx)
+static av_cold int g726_encode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
- unsigned int index;
+ 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, "Samplerate is invalid\n");
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "Invalid sample rate %d\n",
+ avctx->sample_rate);
+ return AVERROR(EINVAL);
}
- index = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate - 2;
-
- if (avctx->bit_rate % avctx->sample_rate && avctx->codec->encode) {
- av_log(avctx, AV_LOG_ERROR, "Bitrate - Samplerate combination is invalid\n");
- return -1;
- }
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
- return -1;
+ return AVERROR(EINVAL);
}
- if(index>3){
- av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
- return -1;
- }
- g726_reset(c, index);
- c->code_size = index+2;
- avctx->coded_frame = avcodec_alloc_frame();
- if (!avctx->coded_frame)
- return AVERROR(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;
- if (avctx->codec->decode)
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ g726_reset(c);
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
- if (avctx->codec->encode)
- avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[index];
+ avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
return 0;
}
-static av_cold int g726_close(AVCodecContext *avctx)
-{
- av_freep(&avctx->coded_frame);
- return 0;
-}
-
-#if CONFIG_ADPCM_G726_ENCODER
-static int g726_encode_frame(AVCodecContext *avctx,
- uint8_t *dst, int buf_size, void *data)
+static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
G726Context *c = avctx->priv_data;
- const short *samples = data;
+ const int16_t *samples = (const int16_t *)frame->data[0];
PutBitContext pb;
- int i;
+ int i, ret, out_size;
- init_put_bits(&pb, dst, 1024*1024);
+ 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 < avctx->frame_size; i++)
+ for (i = 0; i < frame->nb_samples; i++)
put_bits(&pb, c->code_size, g726_encode(c, *samples++));
flush_put_bits(&pb);
- return put_bits_count(&pb)>>3;
+ 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",
+ .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
-static int g726_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+#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;
- short *samples = data;
- GetBitContext gb;
+ int16_t *samples;
+ BitstreamContext bc;
+ int out_samples, ret;
- init_get_bits(&gb, buf, buf_size * 8);
+ out_samples = buf_size * 8 / c->code_size;
- while (get_bits_count(&gb) + c->code_size <= buf_size*8)
- *samples++ = g726_decode(c, get_bits(&gb, 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_init(&bc, buf, buf_size * 8);
+
+ while (out_samples--)
+ *samples++ = g726_decode(c, bitstream_read(&bc, c->code_size));
- if(buf_size*8 != get_bits_count(&gb))
+ if (bitstream_bits_left(&bc) > 0)
av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
- *data_size = (uint8_t*)samples - (uint8_t*)data;
+ *got_frame_ptr = 1;
+
return buf_size;
}
-#if CONFIG_ADPCM_G726_ENCODER
-AVCodec adpcm_g726_encoder = {
- "g726",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_ADPCM_G726,
- sizeof(G726Context),
- g726_init,
- g726_encode_frame,
- g726_close,
- NULL,
- .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"),
-};
-#endif
+static void g726_decode_flush(AVCodecContext *avctx)
+{
+ G726Context *c = avctx->priv_data;
+ g726_reset(c);
+}
-AVCodec adpcm_g726_decoder = {
- "g726",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_ADPCM_G726,
- sizeof(G726Context),
- g726_init,
- NULL,
- g726_close,
- g726_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+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
projects / ffmpeg / blobdiff