#define AMR_AGC_ALPHA 0.9
typedef struct AMRContext {
- AVFrame avframe; ///< AVFrame for decoded samples
AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
enum Mode cur_frame_mode;
int i;
if (avctx->channels > 1) {
- av_log_missing_feature(avctx, "multi-channel AMR", 0);
+ avpriv_report_missing_feature(avctx, "multi-channel AMR");
return AVERROR_PATCHWELCOME;
}
for (i = 0; i < 4; i++)
p->prediction_error[i] = MIN_ENERGY;
- avcodec_get_frame_defaults(&p->avframe);
- avctx->coded_frame = &p->avframe;
-
return 0;
}
* @param p the context
* @param subframe unpacked amr subframe
* @param mode mode of the current frame
- * @param fixed_sparse sparse respresentation of the fixed vector
+ * @param fixed_sparse sparse representation of the fixed vector
*/
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
AMRFixed *fixed_sparse)
{
// The spec suggests the current pitch gain is always used, but in other
- // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
+ // modes the pitch and codebook gains are jointly quantized (sec 5.8.2)
// so the codebook gain cannot depend on the quantized pitch gain.
if (mode == MODE_12k2)
p->beta = FFMIN(p->pitch_gain[4], 1.0);
{
AMRContext *p = avctx->priv_data; // pointer to private data
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
float *buf_out; // pointer to the output data buffer
const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
/* get output buffer */
- p->avframe.nb_samples = AMR_BLOCK_SIZE;
- if ((ret = ff_get_buffer(avctx, &p->avframe)) < 0) {
+ frame->nb_samples = AMR_BLOCK_SIZE;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- buf_out = (float *)p->avframe.data[0];
+ buf_out = (float *)frame->data[0];
p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
if (p->cur_frame_mode == NO_DATA) {
return AVERROR_INVALIDDATA;
}
if (p->cur_frame_mode == MODE_DTX) {
- av_log_missing_feature(avctx, "dtx mode", 1);
+ avpriv_request_sample(avctx, "dtx mode");
return AVERROR_PATCHWELCOME;
}
ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
0.84, 0.16, LP_FILTER_ORDER);
- *got_frame_ptr = 1;
- *(AVFrame *)data = p->avframe;
+ *got_frame_ptr = 1;
/* return the amount of bytes consumed if everything was OK */
return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
AVCodec ff_amrnb_decoder = {
.name = "amrnb",
+ .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AMR_NB,
.priv_data_size = sizeof(AMRContext),
.init = amrnb_decode_init,
.decode = amrnb_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
+ .capabilities = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};