* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-
+
/**
* @file mp3lameaudio.c
* Interface to libmp3lame for mp3 encoding.
#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
typedef struct Mp3AudioContext {
- lame_global_flags *gfp;
- int stereo;
+ lame_global_flags *gfp;
+ int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
} Mp3AudioContext;
static int MP3lame_encode_init(AVCodecContext *avctx)
{
- Mp3AudioContext *s = avctx->priv_data;
-
- if (avctx->channels > 2)
- return -1;
-
- s->stereo = avctx->channels > 1 ? 1 : 0;
-
- if ((s->gfp = lame_init()) == NULL)
- goto err;
- lame_set_in_samplerate(s->gfp, avctx->sample_rate);
- lame_set_out_samplerate(s->gfp, avctx->sample_rate);
- lame_set_num_channels(s->gfp, avctx->channels);
- /* lame 3.91 dies on quality != 5 */
- lame_set_quality(s->gfp, 5);
- /* lame 3.91 doesn't work in mono */
- lame_set_mode(s->gfp, JOINT_STEREO);
- lame_set_brate(s->gfp, avctx->bit_rate/1000);
+ Mp3AudioContext *s = avctx->priv_data;
+
+ if (avctx->channels > 2)
+ return -1;
+
+ s->stereo = avctx->channels > 1 ? 1 : 0;
+
+ if ((s->gfp = lame_init()) == NULL)
+ goto err;
+ lame_set_in_samplerate(s->gfp, avctx->sample_rate);
+ lame_set_out_samplerate(s->gfp, avctx->sample_rate);
+ lame_set_num_channels(s->gfp, avctx->channels);
+ /* lame 3.91 dies on quality != 5 */
+ lame_set_quality(s->gfp, 5);
+ /* lame 3.91 doesn't work in mono */
+ lame_set_mode(s->gfp, JOINT_STEREO);
+ lame_set_brate(s->gfp, avctx->bit_rate/1000);
+ if(avctx->flags & CODEC_FLAG_QSCALE) {
+ lame_set_brate(s->gfp, 0);
+ lame_set_VBR(s->gfp, vbr_default);
+ lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
+ }
lame_set_bWriteVbrTag(s->gfp,0);
- if (lame_init_params(s->gfp) < 0)
- goto err_close;
+ if (lame_init_params(s->gfp) < 0)
+ goto err_close;
+
+ avctx->frame_size = lame_get_framesize(s->gfp);
- avctx->frame_size = MPA_FRAME_SIZE;
-
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
- return 0;
+ return 0;
err_close:
- lame_close(s->gfp);
+ lame_close(s->gfp);
err:
- return -1;
+ return -1;
}
static const int sSampleRates[3] = {
if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
return -1;
}
-
+
if(!samplesPerFrame) samplesPerFrame= &temp0;
if(!sampleRate ) sampleRate = &temp1;
bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
-
+
return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
}
int MP3lame_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
- Mp3AudioContext *s = avctx->priv_data;
- int len, i;
+ Mp3AudioContext *s = avctx->priv_data;
+ int len;
+ int lame_result;
+
+ /* lame 3.91 dies on '1-channel interleaved' data */
- /* lame 3.91 dies on '1-channel interleaved' data */
- if (s->stereo) {
- s->buffer_index += lame_encode_buffer_interleaved(
- s->gfp,
+ if(data){
+ if (s->stereo) {
+ lame_result = lame_encode_buffer_interleaved(
+ s->gfp,
data,
- MPA_FRAME_SIZE,
- s->buffer + s->buffer_index,
+ avctx->frame_size,
+ s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
- } else {
- s->buffer_index += lame_encode_buffer(
- s->gfp,
- data,
- data,
- MPA_FRAME_SIZE,
- s->buffer + 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(s->buffer_index<4)
- return 0;
+ }
+ }else{
+ lame_result= lame_encode_flush(
+ s->gfp,
+ s->buffer + s->buffer_index,
+ BUFFER_SIZE - s->buffer_index
+ );
+ }
+
+ if(lame_result==-1) {
+ /* output buffer too small */
+ av_log(avctx, AV_LOG_ERROR, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s->buffer_index, BUFFER_SIZE - s->buffer_index);
+ return 0;
+ }
+
+ s->buffer_index += lame_result;
+
+ if(s->buffer_index<4)
+ return 0;
len= mp3len(s->buffer, NULL, NULL);
-//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", MPA_FRAME_SIZE, len, s->buffer_index);
+//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
if(len <= s->buffer_index){
memcpy(frame, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer+len, s->buffer_index);
//FIXME fix the audio codec API, so we dont need the memcpy()
- //FIXME fix the audio codec API, so we can output multiple packets if we have them
/*for(i=0; i<len; i++){
av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/
int MP3lame_encode_close(AVCodecContext *avctx)
{
- Mp3AudioContext *s = avctx->priv_data;
-
+ Mp3AudioContext *s = avctx->priv_data;
+
av_freep(&avctx->coded_frame);
- lame_close(s->gfp);
- return 0;
+ lame_close(s->gfp);
+ return 0;
}
sizeof(Mp3AudioContext),
MP3lame_encode_init,
MP3lame_encode_frame,
- MP3lame_encode_close
+ MP3lame_encode_close,
+ .capabilities= CODEC_CAP_DELAY,
};
projects / ffmpeg / blobdiff