*/
#include <limits.h>
-#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.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
+ * numerous occasions. It's a mystery to me why they did it this way
* instead of simply using 32bit integer arithmetic.
*/
typedef struct Float11 {
typedef struct G726Context {
AVClass *class;
- AVFrame frame;
G726Tables tbls; /**< static tables needed for computation */
Float11 sr[2]; /**< prev. reconstructed samples */
g726_reset(c);
-#if FF_API_OLD_ENCODE_AUDIO
- avctx->coded_frame = avcodec_alloc_frame();
- if (!avctx->coded_frame)
- return AVERROR(ENOMEM);
- avctx->coded_frame->key_frame = 1;
-#endif
-
/* 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;
}
-#if FF_API_OLD_ENCODE_AUDIO
-static av_cold int g726_encode_close(AVCodecContext *avctx)
-{
- av_freep(&avctx->coded_frame);
- return 0;
-}
-#endif
-
static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
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,
-#if FF_API_OLD_ENCODE_AUDIO
- .close = g726_encode_close,
-#endif
.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,
};
{
G726Context* c = avctx->priv_data;
- if (avctx->strict_std_compliance >= FF_COMPLIANCE_STRICT &&
- avctx->sample_rate != 8000) {
- av_log(avctx, AV_LOG_ERROR, "Only 8kHz sample rate is allowed when "
- "the compliance level is strict. Reduce the compliance level "
- "if you wish to decode the stream anyway.\n");
- return AVERROR(EINVAL);
- }
-
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- avcodec_get_frame_defaults(&c->frame);
- avctx->coded_frame = &c->frame;
-
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;
out_samples = buf_size * 8 / c->code_size;
/* get output buffer */
- c->frame.nb_samples = out_samples;
- if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
+ 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 *)c->frame.data[0];
+ samples = (int16_t *)frame->data[0];
init_get_bits(&gb, buf, buf_size * 8);
if (get_bits_left(&gb) > 0)
av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
- *got_frame_ptr = 1;
- *(AVFrame *)data = c->frame;
+ *got_frame_ptr = 1;
return buf_size;
}
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),
.decode = g726_decode_frame,
.flush = g726_decode_flush,
.capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
};
#endif