/*
* copyright (c) 2001 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/apiexample.c
- * avcodec API use example.
+ * @file
+ * libavcodec API use example.
*
+ * @example libavcodec/api-example.c
* Note that this library only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, etc...). See library 'libavformat' for the
* format handling
#undef HAVE_AV_CONFIG_H
#endif
-#include "avcodec.h"
+#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"
#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
/*
* Audio encoding example
exit(1);
}
- c= avcodec_alloc_context();
+ c = avcodec_alloc_context3(codec);
/* put sample parameters */
c->bit_rate = 64000;
{
AVCodec *codec;
AVCodecContext *c= NULL;
- int out_size, size, len;
+ int len;
FILE *f, *outfile;
- uint8_t *outbuf;
- uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE], *inbuf_ptr;
+ uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+ AVPacket avpkt;
+ AVFrame *decoded_frame = NULL;
+
+ av_init_packet(&avpkt);
printf("Audio decoding\n");
exit(1);
}
- c= avcodec_alloc_context();
+ c = avcodec_alloc_context3(codec);
/* open it */
if (avcodec_open(c, codec) < 0) {
exit(1);
}
- outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
-
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
}
/* decode until eof */
- inbuf_ptr = inbuf;
- for(;;) {
- size = fread(inbuf, 1, INBUF_SIZE, f);
- if (size == 0)
- break;
+ avpkt.data = inbuf;
+ avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
- inbuf_ptr = inbuf;
- while (size > 0) {
- out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- len = avcodec_decode_audio2(c, (short *)outbuf, &out_size,
- inbuf_ptr, size);
- if (len < 0) {
- fprintf(stderr, "Error while decoding\n");
+ while (avpkt.size > 0) {
+ int got_frame = 0;
+
+ if (!decoded_frame) {
+ if (!(decoded_frame = avcodec_alloc_frame())) {
+ fprintf(stderr, "out of memory\n");
exit(1);
}
- if (out_size > 0) {
- /* if a frame has been decoded, output it */
- fwrite(outbuf, 1, out_size, outfile);
- }
- size -= len;
- inbuf_ptr += len;
+ } else
+ avcodec_get_frame_defaults(decoded_frame);
+
+ len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+ if (len < 0) {
+ fprintf(stderr, "Error while decoding\n");
+ exit(1);
+ }
+ if (got_frame) {
+ /* if a frame has been decoded, output it */
+ int data_size = av_samples_get_buffer_size(NULL, c->channels,
+ decoded_frame->nb_samples,
+ c->sample_fmt, 1);
+ fwrite(decoded_frame->data[0], 1, data_size, outfile);
+ }
+ avpkt.size -= len;
+ avpkt.data += len;
+ if (avpkt.size < AUDIO_REFILL_THRESH) {
+ /* Refill the input buffer, to avoid trying to decode
+ * incomplete frames. Instead of this, one could also use
+ * a parser, or use a proper container format through
+ * libavformat. */
+ memmove(inbuf, avpkt.data, avpkt.size);
+ avpkt.data = inbuf;
+ len = fread(avpkt.data + avpkt.size, 1,
+ AUDIO_INBUF_SIZE - avpkt.size, f);
+ if (len > 0)
+ avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
- free(outbuf);
avcodec_close(c);
av_free(c);
+ av_free(decoded_frame);
}
/*
exit(1);
}
- c= avcodec_alloc_context();
+ c = avcodec_alloc_context3(codec);
picture= avcodec_alloc_frame();
/* put sample parameters */
{
AVCodec *codec;
AVCodecContext *c= NULL;
- int frame, size, got_picture, len;
+ int frame, got_picture, len;
FILE *f;
AVFrame *picture;
- uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE], *inbuf_ptr;
+ uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
char buf[1024];
+ AVPacket avpkt;
+
+ av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
exit(1);
}
- c= avcodec_alloc_context();
+ c = avcodec_alloc_context3(codec);
picture= avcodec_alloc_frame();
if(codec->capabilities&CODEC_CAP_TRUNCATED)
frame = 0;
for(;;) {
- size = fread(inbuf, 1, INBUF_SIZE, f);
- if (size == 0)
+ avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+ if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
- inbuf_ptr = inbuf;
- while (size > 0) {
- len = avcodec_decode_video(c, picture, &got_picture,
- inbuf_ptr, size);
+ avpkt.data = inbuf;
+ while (avpkt.size > 0) {
+ len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", frame);
exit(1);
c->width, c->height, buf);
frame++;
}
- size -= len;
- inbuf_ptr += len;
+ avpkt.size -= len;
+ avpkt.data += len;
}
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
- len = avcodec_decode_video(c, picture, &got_picture,
- NULL, 0);
+ avpkt.data = NULL;
+ avpkt.size = 0;
+ len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (got_picture) {
printf("saving last frame %3d\n", frame);
fflush(stdout);
projects / ffmpeg / blobdiff