* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
+ struct {
+ int *left;
+ int *right;
+ } s32_data;
int reservoir;
} Mp3AudioContext;
{
Mp3AudioContext *s = avctx->priv_data;
- if (avctx->channels > 2)
- return -1;
+ if (avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Invalid number of channels %d, must be <= 2\n", avctx->channels);
+ return AVERROR(EINVAL);
+ }
s->stereo = avctx->channels > 1 ? 1 : 0;
lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
}
lame_set_bWriteVbrTag(s->gfp,0);
-#if FF_API_LAME_GLOBAL_OPTIONS
+#if FF_API_LAME_GLOBAL_OPTS
s->reservoir = avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR;
#endif
lame_set_disable_reservoir(s->gfp, !s->reservoir);
avctx->frame_size = lame_get_framesize(s->gfp);
- avctx->coded_frame= avcodec_alloc_frame();
+ if(!(avctx->coded_frame= avcodec_alloc_frame())) {
+ lame_close(s->gfp);
+
+ return AVERROR(ENOMEM);
+ }
avctx->coded_frame->key_frame= 1;
+ if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
+ int nelem = 2 * avctx->frame_size;
+
+ if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
+ av_freep(&avctx->coded_frame);
+ lame_close(s->gfp);
+
+ return AVERROR(ENOMEM);
+ }
+
+ s->s32_data.right = s->s32_data.left + avctx->frame_size;
+ }
+
return 0;
err_close:
/* lame 3.91 dies on '1-channel interleaved' data */
- if(data){
+ if(!data){
+ lame_result= lame_encode_flush(
+ s->gfp,
+ s->buffer + s->buffer_index,
+ BUFFER_SIZE - s->buffer_index
+ );
+#if 2147483647 == INT_MAX
+ }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
if (s->stereo) {
- lame_result = lame_encode_buffer_interleaved(
+ int32_t *rp = data;
+ int32_t *mp = rp + 2*avctx->frame_size;
+ int *wpl = s->s32_data.left;
+ int *wpr = s->s32_data.right;
+
+ while (rp < mp) {
+ *wpl++ = *rp++;
+ *wpr++ = *rp++;
+ }
+
+ lame_result = lame_encode_buffer_int(
s->gfp,
- data,
+ s->s32_data.left,
+ s->s32_data.right,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} else {
- lame_result = lame_encode_buffer(
+ lame_result = lame_encode_buffer_int(
s->gfp,
data,
data,
BUFFER_SIZE - s->buffer_index
);
}
+#endif
}else{
- lame_result= lame_encode_flush(
+ if (s->stereo) {
+ lame_result = lame_encode_buffer_interleaved(
+ s->gfp,
+ data,
+ avctx->frame_size,
+ s->buffer + s->buffer_index,
+ BUFFER_SIZE - s->buffer_index
+ );
+ } else {
+ lame_result = lame_encode_buffer(
s->gfp,
+ data,
+ data,
+ avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
+ }
}
if(lame_result < 0){
{
Mp3AudioContext *s = avctx->priv_data;
+ av_freep(&s->s32_data.left);
av_freep(&avctx->coded_frame);
lame_close(s->gfp);
#define OFFSET(x) offsetof(Mp3AudioContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
- { "reservoir", "Use bit reservoir.", OFFSET(reservoir), FF_OPT_TYPE_INT, { 1 }, 0, 1, AE },
+ { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
{ NULL },
};
.encode = MP3lame_encode_frame,
.close = MP3lame_encode_close,
.capabilities= CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
+#if 2147483647 == INT_MAX
+ AV_SAMPLE_FMT_S32,
+#endif
+ AV_SAMPLE_FMT_NONE},
.supported_samplerates= sSampleRates,
.long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,