1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-21 10:55:51 +02:00

flacenc: use AVCodec.encode2()

This commit is contained in:
Justin Ruggles 2012-02-03 18:43:39 -05:00
parent 24e74f0a0f
commit 910bdb9a42

View File

@ -25,6 +25,7 @@
#include "avcodec.h"
#include "get_bits.h"
#include "golomb.h"
#include "internal.h"
#include "lpc.h"
#include "flac.h"
#include "flacdata.h"
@ -367,9 +368,11 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->frame_count = 0;
s->min_framesize = s->max_framesize;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON);
@ -380,7 +383,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
}
static void init_frame(FlacEncodeContext *s)
static void init_frame(FlacEncodeContext *s, int nb_samples)
{
int i, ch;
FlacFrame *frame;
@ -388,7 +391,7 @@ static void init_frame(FlacEncodeContext *s)
frame = &s->frame;
for (i = 0; i < 16; i++) {
if (s->avctx->frame_size == ff_flac_blocksize_table[i]) {
if (nb_samples == ff_flac_blocksize_table[i]) {
frame->blocksize = ff_flac_blocksize_table[i];
frame->bs_code[0] = i;
frame->bs_code[1] = 0;
@ -396,7 +399,7 @@ static void init_frame(FlacEncodeContext *s)
}
}
if (i == 16) {
frame->blocksize = s->avctx->frame_size;
frame->blocksize = nb_samples;
if (frame->blocksize <= 256) {
frame->bs_code[0] = 6;
frame->bs_code[1] = frame->blocksize-1;
@ -1166,9 +1169,9 @@ static void write_frame_footer(FlacEncodeContext *s)
}
static int write_frame(FlacEncodeContext *s, uint8_t *frame, int buf_size)
static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
{
init_put_bits(&s->pb, frame, buf_size);
init_put_bits(&s->pb, avpkt->data, avpkt->size);
write_frame_header(s);
write_subframes(s);
write_frame_footer(s);
@ -1190,30 +1193,31 @@ static void update_md5_sum(FlacEncodeContext *s, const int16_t *samples)
}
static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
FlacEncodeContext *s;
const int16_t *samples = data;
int frame_bytes, out_bytes;
const int16_t *samples;
int frame_bytes, out_bytes, ret;
s = avctx->priv_data;
/* when the last block is reached, update the header in extradata */
if (!data) {
if (!frame) {
s->max_framesize = s->max_encoded_framesize;
av_md5_final(s->md5ctx, s->md5sum);
write_streaminfo(s, avctx->extradata);
return 0;
}
samples = (const int16_t *)frame->data[0];
/* change max_framesize for small final frame */
if (avctx->frame_size < s->frame.blocksize) {
s->max_framesize = ff_flac_get_max_frame_size(avctx->frame_size,
if (frame->nb_samples < s->frame.blocksize) {
s->max_framesize = ff_flac_get_max_frame_size(frame->nb_samples,
s->channels, 16);
}
init_frame(s);
init_frame(s, frame->nb_samples);
copy_samples(s, samples);
@ -1228,22 +1232,26 @@ static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
frame_bytes = encode_frame(s);
}
if (buf_size < frame_bytes) {
av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
return 0;
if ((ret = ff_alloc_packet(avpkt, frame_bytes))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
out_bytes = write_frame(s, frame, buf_size);
out_bytes = write_frame(s, avpkt);
s->frame_count++;
avctx->coded_frame->pts = s->sample_count;
s->sample_count += avctx->frame_size;
s->sample_count += frame->nb_samples;
update_md5_sum(s, samples);
if (out_bytes > s->max_encoded_framesize)
s->max_encoded_framesize = out_bytes;
if (out_bytes < s->min_framesize)
s->min_framesize = out_bytes;
return out_bytes;
avpkt->pts = frame->pts;
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
avpkt->size = out_bytes;
*got_packet_ptr = 1;
return 0;
}
@ -1256,7 +1264,9 @@ static av_cold int flac_encode_close(AVCodecContext *avctx)
}
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
@ -1294,7 +1304,7 @@ AVCodec ff_flac_encoder = {
.id = CODEC_ID_FLAC,
.priv_data_size = sizeof(FlacEncodeContext),
.init = flac_encode_init,
.encode = flac_encode_frame,
.encode2 = flac_encode_frame,
.close = flac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},