* 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
*/
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
-#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
+#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
typedef struct LAMEContext {
AVClass *class;
int reservoir;
int joint_stereo;
int abr;
+ int delay_sent;
float *samples_flt[2];
AudioFrameQueue afq;
- AVFloatDSPContext fdsp;
+ AVFloatDSPContext *fdsp;
} LAMEContext;
av_freep(&s->samples_flt[0]);
av_freep(&s->samples_flt[1]);
av_freep(&s->buffer);
+ av_freep(&s->fdsp);
ff_af_queue_close(&s->afq);
if (!(s->gfp = lame_init()))
return AVERROR(ENOMEM);
+
lame_set_num_channels(s->gfp, avctx->channels);
lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
lame_set_out_samplerate(s->gfp, avctx->sample_rate);
/* algorithmic quality */
- if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
- lame_set_quality(s->gfp, 5);
- else
+ if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
lame_set_quality(s->gfp, avctx->compression_level);
/* rate control */
}
}
+ /* lowpass cutoff frequency */
+ if (avctx->cutoff)
+ lame_set_lowpassfreq(s->gfp, avctx->cutoff);
+
/* do not get a Xing VBR header frame from LAME */
lame_set_bWriteVbrTag(s->gfp,0);
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
int ch;
for (ch = 0; ch < avctx->channels; ch++) {
- s->samples_flt[ch] = av_malloc(avctx->frame_size *
+ s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
sizeof(*s->samples_flt[ch]));
if (!s->samples_flt[ch]) {
ret = AVERROR(ENOMEM);
if (ret < 0)
goto error;
- avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ if (!s->fdsp) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+
return 0;
error:
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
- int len, ret, ch;
+ int len, ret, ch, discard_padding;
int lame_result;
uint32_t h;
return AVERROR(EINVAL);
}
for (ch = 0; ch < avctx->channels; ch++) {
- s->fdsp.vector_fmul_scalar(s->samples_flt[ch],
+ s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
(const float *)frame->data[ch],
32768.0f,
FFALIGN(frame->nb_samples, 8));
default:
return AVERROR_BUG;
}
+ } else if (!s->afq.frame_alloc) {
+ lame_result = 0;
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
s->buffer_size - s->buffer_index);
ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
s->buffer_index);
if (len <= s->buffer_index) {
- if ((ret = ff_alloc_packet(avpkt, len))) {
- av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
return ret;
- }
memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
+ discard_padding = avctx->frame_size - avpkt->duration;
+ // Check if subtraction resulted in an overflow
+ if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
+ av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
+ av_packet_unref(avpkt);
+ av_free(avpkt);
+ return AVERROR(EINVAL);
+ }
+ if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
+ uint8_t* side_data = av_packet_new_side_data(avpkt,
+ AV_PKT_DATA_SKIP_SAMPLES,
+ 10);
+ if(!side_data) {
+ av_packet_unref(avpkt);
+ av_free(avpkt);
+ return AVERROR(ENOMEM);
+ }
+ if (!s->delay_sent) {
+ AV_WL32(side_data, avctx->initial_padding);
+ s->delay_sent = 1;
+ }
+ AV_WL32(side_data + 4, discard_padding);
+ }
+
avpkt->size = len;
*got_packet_ptr = 1;
}
#define OFFSET(x) offsetof(LAMEContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
- { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
- { "joint_stereo", "Use joint stereo.", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
- { "abr", "Use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
+ { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
+ { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
+ { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
{ NULL },
};
projects / ffmpeg / blobdiff