* The simplest mpeg audio layer 2 encoder.
*/
+#include "libavutil/audioconvert.h"
+
#include "avcodec.h"
#include "internal.h"
#include "put_bits.h"
#define WFRAC_BITS 14 /* fractional bits for window */
#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
/* currently, cannot change these constants (need to modify
quantization stage) */
if (channels <= 0 || channels > 2){
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
- return -1;
+ return AVERROR(EINVAL);
}
bitrate = bitrate / 1000;
s->nb_channels = channels;
avctx->frame_size = MPA_FRAME_SIZE;
+ avctx->delay = 512 - 32 + 1;
/* encoding freq */
s->lsf = 0;
}
if (i == 3){
av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
- return -1;
+ return AVERROR(EINVAL);
}
s->freq_index = i;
}
if (i == 15){
av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
- return -1;
+ return AVERROR(EINVAL);
}
s->bitrate_index = i;
total_quant_bits[i] = 12 * v;
}
+#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
- avctx->coded_frame->key_frame= 1;
+ if (!avctx->coded_frame)
+ return AVERROR(ENOMEM);
+#endif
return 0;
}
flush_put_bits(p);
}
-static int MPA_encode_frame(AVCodecContext *avctx,
- unsigned char *frame, int buf_size, void *data)
+static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
MpegAudioContext *s = avctx->priv_data;
- const short *samples = data;
+ const int16_t *samples = (const int16_t *)frame->data[0];
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
- int padding, i;
+ int padding, i, ret;
for(i=0;i<s->nb_channels;i++) {
filter(s, i, samples + i, s->nb_channels);
}
compute_bit_allocation(s, smr, bit_alloc, &padding);
- init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
+ if ((ret = ff_alloc_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+
+ init_put_bits(&s->pb, avpkt->data, avpkt->size);
encode_frame(s, bit_alloc, padding);
- return put_bits_ptr(&s->pb) - s->pb.buf;
+ if (frame->pts != AV_NOPTS_VALUE)
+ avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+
+ avpkt->size = put_bits_count(&s->pb) / 8;
+ *got_packet_ptr = 1;
+ return 0;
}
static av_cold int MPA_encode_close(AVCodecContext *avctx)
{
+#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
+#endif
return 0;
}
};
AVCodec ff_mp2_encoder = {
- .name = "mp2",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_MP2,
- .priv_data_size = sizeof(MpegAudioContext),
- .init = MPA_encode_init,
- .encode = MPA_encode_frame,
- .close = MPA_encode_close,
- .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)"),
- .defaults = mp2_defaults,
+ .name = "mp2",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_MP2,
+ .priv_data_size = sizeof(MpegAudioContext),
+ .init = MPA_encode_init,
+ .encode2 = MPA_encode_frame,
+ .close = MPA_encode_close,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .supported_samplerates = (const int[]){
+ 44100, 48000, 32000, 22050, 24000, 16000, 0
+ },
+ .channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ 0 },
+ .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
+ .defaults = mp2_defaults,
};