/*
* The simplest mpeg audio layer 2 encoder
- * Copyright (c) 2000, 2001 Fabrice Bellard.
+ * Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
*/
/**
- * @file mpegaudio.c
+ * @file
* The simplest mpeg audio layer 2 encoder.
*/
#include "avcodec.h"
-#include "bitstream.h"
+#include "put_bits.h"
+
+#undef CONFIG_MPEGAUDIO_HP
+#define CONFIG_MPEGAUDIO_HP 0
#include "mpegaudio.h"
/* currently, cannot change these constants (need to modify
quantization stage) */
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
-#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
#define SAMPLES_BUF_SIZE 4096
typedef struct MpegAudioContext {
PutBitContext pb;
int nb_channels;
- int freq, bit_rate;
int lsf; /* 1 if mpeg2 low bitrate selected */
int bitrate_index; /* bit rate */
int freq_index;
int frame_size; /* frame size, in bits, without padding */
- int64_t nb_samples; /* total number of samples encoded */
/* padding computation */
int frame_frac, frame_frac_incr, do_padding;
short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
} MpegAudioContext;
/* define it to use floats in quantization (I don't like floats !) */
-//#define USE_FLOATS
+#define USE_FLOATS
#include "mpegaudiodata.h"
#include "mpegaudiotab.h"
-static int MPA_encode_init(AVCodecContext *avctx)
+static av_cold int MPA_encode_init(AVCodecContext *avctx)
{
MpegAudioContext *s = avctx->priv_data;
int freq = avctx->sample_rate;
}
bitrate = bitrate / 1000;
s->nb_channels = channels;
- s->freq = freq;
- s->bit_rate = bitrate * 1000;
avctx->frame_size = MPA_FRAME_SIZE;
/* encoding freq */
s->sblimit = ff_mpa_sblimit_table[table];
s->alloc_table = ff_mpa_alloc_tables[table];
-#ifdef DEBUG
- av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
- bitrate, freq, s->frame_size, table, s->frame_frac_incr);
-#endif
+ av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
+ bitrate, freq, s->frame_size, table, s->frame_frac_incr);
for(i=0;i<s->nb_channels;i++)
s->samples_offset[i] = 0;
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
-static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
+static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
{
short *p, *q;
int sum, offset, i, j;
vmax = v;
}
/* compute the scale factor index using log 2 computations */
- if (vmax > 0) {
+ if (vmax > 1) {
n = av_log2(vmax);
/* n is the position of the MSB of vmax. now
use at most 2 compares to find the index */
sf[1] = sf[2] = sf[0];
break;
default:
- assert(0); //cant happen
+ assert(0); //cannot happen
code = 0; /* kill warning */
}
/* look for the subband with the largest signal to mask ratio */
max_sb = -1;
max_ch = -1;
- max_smr = 0x80000000;
+ max_smr = INT_MIN;
for(ch=0;ch<s->nb_channels;ch++) {
for(i=0;i<s->sblimit;i++) {
if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
unsigned char *frame, int buf_size, void *data)
{
MpegAudioContext *s = avctx->priv_data;
- short *samples = data;
+ const short *samples = data;
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
int padding, i;
encode_frame(s, bit_alloc, padding);
- s->nb_samples += MPA_FRAME_SIZE;
- return pbBufPtr(&s->pb) - s->pb.buf;
+ return put_bits_ptr(&s->pb) - s->pb.buf;
}
-static int MPA_encode_close(AVCodecContext *avctx)
+static av_cold int MPA_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
-AVCodec mp2_encoder = {
+AVCodec ff_mp2_encoder = {
"mp2",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_MP2,
sizeof(MpegAudioContext),
MPA_encode_init,
MPA_encode_frame,
MPA_encode_close,
NULL,
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0},
+ .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
};
#undef FIX