* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
- * 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 mp3lameaudio.c
+ * @file
* Interface to libmp3lame for mp3 encoding.
*/
+#include "libavutil/intreadwrite.h"
+#include "libavutil/log.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
#include "mpegaudio.h"
#include <lame/lame.h>
-#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
+#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
typedef struct Mp3AudioContext {
+ AVClass *class;
lame_global_flags *gfp;
int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
+ int reservoir;
} Mp3AudioContext;
-static int MP3lame_encode_init(AVCodecContext *avctx)
+static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
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) {
+ if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+ lame_set_quality(s->gfp, 5);
+ } else {
+ lame_set_quality(s->gfp, avctx->compression_level);
+ }
+ lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
+ 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_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
}
lame_set_bWriteVbrTag(s->gfp,0);
+#if FF_API_LAME_GLOBAL_OPTS
+ s->reservoir = avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR;
+#endif
+ lame_set_disable_reservoir(s->gfp, !s->reservoir);
if (lame_init_params(s->gfp) < 0)
goto err_close;
- avctx->frame_size = lame_get_framesize(s->gfp);
-
- avctx->coded_frame= avcodec_alloc_frame();
- avctx->coded_frame->key_frame= 1;
+ avctx->frame_size = lame_get_framesize(s->gfp);
+ avctx->coded_frame = avcodec_alloc_frame();
+ avctx->coded_frame->key_frame = 1;
return 0;
return -1;
}
-static const int sSampleRates[3] = {
- 44100, 48000, 32000
+static const int sSampleRates[] = {
+ 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
static const int sBitRates[2][3][15] = {
- { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
- { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
- { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
+ {
+ { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
+ { 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
+ { 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
},
- { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
- { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
- { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
+ {
+ { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
+ { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
+ { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
},
};
-static const int sSamplesPerFrame[2][3] =
-{
- { 384, 1152, 1152 },
- { 384, 1152, 576 }
+static const int sSamplesPerFrame[2][3] = {
+ { 384, 1152, 1152 },
+ { 384, 1152, 576 }
};
-static const int sBitsPerSlot[3] = {
- 32,
- 8,
- 8
-};
+static const int sBitsPerSlot[3] = { 32, 8, 8 };
static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
{
- uint32_t header = AV_RB32(data);
- int layerID = 3 - ((header >> 17) & 0x03);
- int bitRateID = ((header >> 12) & 0x0f);
+ uint32_t header = AV_RB32(data);
+ int layerID = 3 - ((header >> 17) & 0x03);
+ int bitRateID = ((header >> 12) & 0x0f);
int sampleRateID = ((header >> 10) & 0x03);
- int bitsPerSlot = sBitsPerSlot[layerID];
- int isPadded = ((header >> 9) & 0x01);
- static int const mode_tab[4]= {2,3,1,0};
- int mode= mode_tab[(header >> 19) & 0x03];
- int mpeg_id= mode>0;
+ int bitsPerSlot = sBitsPerSlot[layerID];
+ int isPadded = ((header >> 9) & 0x01);
+ static int const mode_tab[4] = { 2, 3, 1, 0 };
+ int mode = mode_tab[(header >> 19) & 0x03];
+ int mpeg_id = mode > 0;
int temp0, temp1, bitRate;
- if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
+ if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
+ sampleRateID == 3) {
return -1;
}
- if(!samplesPerFrame) samplesPerFrame= &temp0;
- if(!sampleRate ) sampleRate = &temp1;
+ if (!samplesPerFrame)
+ samplesPerFrame = &temp0;
+ if (!sampleRate)
+ sampleRate = &temp1;
-// *isMono = ((header >> 6) & 0x03) == 0x03;
+ //*isMono = ((header >> 6) & 0x03) == 0x03;
- *sampleRate = sSampleRates[sampleRateID]>>mode;
- bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
+ *sampleRate = sSampleRates[sampleRateID] >> mode;
+ 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);
+ //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;
}
-static int MP3lame_encode_frame(AVCodecContext *avctx,
- unsigned char *frame, int buf_size, void *data)
+static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+ int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
int len;
/* lame 3.91 dies on '1-channel interleaved' data */
- if(data){
+ if (data) {
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
- );
+ 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
- );
+ lame_result = lame_encode_buffer(s->gfp, data, data,
+ avctx->frame_size, s->buffer +
+ s->buffer_index, BUFFER_SIZE -
+ s->buffer_index);
}
- }else{
- lame_result= lame_encode_flush(
- s->gfp,
- s->buffer + s->buffer_index,
- BUFFER_SIZE - s->buffer_index
- );
+ } 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;
+ if (lame_result < 0) {
+ 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 -1;
}
s->buffer_index += lame_result;
- if(s->buffer_index<4)
+ 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", 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 do not need the memcpy()
-/*for(i=0; i<len; i++){
- av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
-}*/
- return len;
- }else
- return 0;
+ len = mp3len(s->buffer, NULL, NULL);
+ //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 do not need the memcpy()
+ /*for(i=0; i<len; i++) {
+ av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
+ }*/
+ return len;
+ } else
+ return 0;
}
-static int MP3lame_encode_close(AVCodecContext *avctx)
+static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
return 0;
}
+#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), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
+ { NULL },
+};
+
+static const AVClass libmp3lame_class = {
+ .class_name = "libmp3lame encoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
-AVCodec libmp3lame_encoder = {
- "libmp3lame",
- CODEC_TYPE_AUDIO,
- CODEC_ID_MP3,
- sizeof(Mp3AudioContext),
- MP3lame_encode_init,
- MP3lame_encode_frame,
- MP3lame_encode_close,
- .capabilities= CODEC_CAP_DELAY,
+AVCodec ff_libmp3lame_encoder = {
+ .name = "libmp3lame",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_MP3,
+ .priv_data_size = sizeof(Mp3AudioContext),
+ .init = MP3lame_encode_init,
+ .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 },
+ .supported_samplerates = sSampleRates,
+ .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+ .priv_class = &libmp3lame_class,
};
projects / ffmpeg / blobdiff