FFmpeg/libavcodec/libmp3lame.c
Anton Khirnov 2df0c32ea1 lavc: use a separate field for exporting audio encoder padding
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
  meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
  codec context is used for encoding or decoding (and has yet another
  different meaning for video), preventing generic handling of the codec
  context.

Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
2014-10-13 19:09:01 +00:00

318 lines
10 KiB
C

/*
* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
* This file is part of Libav.
*
* 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.
*
* 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Interface to libmp3lame for mp3 encoding.
*/
#include <lame/lame.h>
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
typedef struct LAMEContext {
AVClass *class;
AVCodecContext *avctx;
lame_global_flags *gfp;
uint8_t *buffer;
int buffer_index;
int buffer_size;
int reservoir;
int joint_stereo;
int abr;
float *samples_flt[2];
AudioFrameQueue afq;
AVFloatDSPContext fdsp;
} LAMEContext;
static int realloc_buffer(LAMEContext *s)
{
if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
new_size);
if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
s->buffer_size = s->buffer_index = 0;
return err;
}
s->buffer_size = new_size;
}
return 0;
}
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
av_freep(&s->samples_flt[0]);
av_freep(&s->samples_flt[1]);
av_freep(&s->buffer);
ff_af_queue_close(&s->afq);
lame_close(s->gfp);
return 0;
}
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
int ret;
s->avctx = avctx;
/* initialize LAME and get defaults */
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);
/* sample rate */
lame_set_in_samplerate (s->gfp, avctx->sample_rate);
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
lame_set_quality(s->gfp, avctx->compression_level);
/* rate control */
if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
lame_set_VBR(s->gfp, vbr_default);
lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
} else {
if (avctx->bit_rate) {
if (s->abr) { // ABR
lame_set_VBR(s->gfp, vbr_abr);
lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
} else // CBR
lame_set_brate(s->gfp, avctx->bit_rate / 1000);
}
}
/* do not get a Xing VBR header frame from LAME */
lame_set_bWriteVbrTag(s->gfp,0);
/* bit reservoir usage */
lame_set_disable_reservoir(s->gfp, !s->reservoir);
/* set specified parameters */
if (lame_init_params(s->gfp) < 0) {
ret = -1;
goto error;
}
/* get encoder delay */
avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
ff_af_queue_init(avctx, &s->afq);
avctx->frame_size = lame_get_framesize(s->gfp);
/* allocate float sample buffers */
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 *
sizeof(*s->samples_flt[ch]));
if (!s->samples_flt[ch]) {
ret = AVERROR(ENOMEM);
goto error;
}
}
}
ret = realloc_buffer(s);
if (ret < 0)
goto error;
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
return 0;
error:
mp3lame_encode_close(avctx);
return ret;
}
#define ENCODE_BUFFER(func, buf_type, buf_name) do { \
lame_result = func(s->gfp, \
(const buf_type *)buf_name[0], \
(const buf_type *)buf_name[1], frame->nb_samples, \
s->buffer + s->buffer_index, \
s->buffer_size - s->buffer_index); \
} while (0)
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len, ret, ch;
int lame_result;
uint32_t h;
if (frame) {
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16P:
ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
break;
case AV_SAMPLE_FMT_S32P:
ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
break;
case AV_SAMPLE_FMT_FLTP:
if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
return AVERROR(EINVAL);
}
for (ch = 0; ch < avctx->channels; ch++) {
s->fdsp.vector_fmul_scalar(s->samples_flt[ch],
(const float *)frame->data[ch],
32768.0f,
FFALIGN(frame->nb_samples, 8));
}
ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
break;
default:
return AVERROR_BUG;
}
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
s->buffer_size - s->buffer_index);
}
if (lame_result < 0) {
if (lame_result == -1) {
av_log(avctx, AV_LOG_ERROR,
"lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
s->buffer_index, s->buffer_size - s->buffer_index);
}
return -1;
}
s->buffer_index += lame_result;
ret = realloc_buffer(s);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
return ret;
}
/* add current frame to the queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
}
/* Move 1 frame from the LAME buffer to the output packet, if available.
We have to parse the first frame header in the output buffer to
determine the frame size. */
if (s->buffer_index < 4)
return 0;
h = AV_RB32(s->buffer);
if (ff_mpa_check_header(h) < 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
return AVERROR_BUG;
}
if (avpriv_mpegaudio_decode_header(&hdr, h)) {
av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
return -1;
}
len = hdr.frame_size;
av_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");
return ret;
}
memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = len;
*got_packet_ptr = 1;
}
return 0;
}
#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 },
{ NULL },
};
static const AVClass libmp3lame_class = {
.class_name = "libmp3lame encoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault libmp3lame_defaults[] = {
{ "b", "0" },
{ NULL },
};
static const int libmp3lame_sample_rates[] = {
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
AVCodec ff_libmp3lame_encoder = {
.name = "libmp3lame",
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(LAMEContext),
.init = mp3lame_encode_init,
.encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = libmp3lame_sample_rates,
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
0 },
.priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults,
};