forked from KolibriOS/kolibrios
308 lines
10 KiB
C
308 lines
10 KiB
C
/*
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* Interface to libmp3lame for mp3 encoding
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* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Interface to libmp3lame for mp3 encoding.
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*/
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#include <lame/lame.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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#include "mpegaudio.h"
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#include "mpegaudiodecheader.h"
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#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.
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typedef struct LAMEContext {
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AVClass *class;
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AVCodecContext *avctx;
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lame_global_flags *gfp;
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uint8_t *buffer;
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int buffer_index;
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int buffer_size;
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int reservoir;
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int joint_stereo;
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float *samples_flt[2];
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AudioFrameQueue afq;
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AVFloatDSPContext fdsp;
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} LAMEContext;
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static int realloc_buffer(LAMEContext *s)
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{
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if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
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uint8_t *tmp;
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int new_size = s->buffer_index + 2 * BUFFER_SIZE;
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av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
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new_size);
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tmp = av_realloc(s->buffer, new_size);
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if (!tmp) {
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av_freep(&s->buffer);
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s->buffer_size = s->buffer_index = 0;
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return AVERROR(ENOMEM);
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}
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s->buffer = tmp;
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s->buffer_size = new_size;
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}
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return 0;
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}
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static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
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{
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LAMEContext *s = avctx->priv_data;
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av_freep(&s->samples_flt[0]);
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av_freep(&s->samples_flt[1]);
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av_freep(&s->buffer);
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ff_af_queue_close(&s->afq);
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lame_close(s->gfp);
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return 0;
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}
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static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
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{
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LAMEContext *s = avctx->priv_data;
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int ret;
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s->avctx = avctx;
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/* initialize LAME and get defaults */
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if ((s->gfp = lame_init()) == NULL)
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return AVERROR(ENOMEM);
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lame_set_num_channels(s->gfp, avctx->channels);
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lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
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/* sample rate */
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lame_set_in_samplerate (s->gfp, avctx->sample_rate);
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lame_set_out_samplerate(s->gfp, avctx->sample_rate);
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/* algorithmic quality */
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if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
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lame_set_quality(s->gfp, 5);
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else
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lame_set_quality(s->gfp, avctx->compression_level);
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/* rate control */
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if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
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lame_set_VBR(s->gfp, vbr_default);
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lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
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} else {
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if (avctx->bit_rate) // CBR
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lame_set_brate(s->gfp, avctx->bit_rate / 1000);
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}
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/* do not get a Xing VBR header frame from LAME */
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lame_set_bWriteVbrTag(s->gfp,0);
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/* bit reservoir usage */
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lame_set_disable_reservoir(s->gfp, !s->reservoir);
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/* set specified parameters */
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if (lame_init_params(s->gfp) < 0) {
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ret = -1;
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goto error;
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}
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/* get encoder delay */
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avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
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ff_af_queue_init(avctx, &s->afq);
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avctx->frame_size = lame_get_framesize(s->gfp);
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/* allocate float sample buffers */
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
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int ch;
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for (ch = 0; ch < avctx->channels; ch++) {
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s->samples_flt[ch] = av_malloc(avctx->frame_size *
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sizeof(*s->samples_flt[ch]));
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if (!s->samples_flt[ch]) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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}
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}
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ret = realloc_buffer(s);
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if (ret < 0)
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goto error;
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avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
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return 0;
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error:
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mp3lame_encode_close(avctx);
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return ret;
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}
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#define ENCODE_BUFFER(func, buf_type, buf_name) do { \
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lame_result = func(s->gfp, \
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(const buf_type *)buf_name[0], \
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(const buf_type *)buf_name[1], frame->nb_samples, \
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s->buffer + s->buffer_index, \
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s->buffer_size - s->buffer_index); \
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} while (0)
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static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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LAMEContext *s = avctx->priv_data;
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MPADecodeHeader hdr;
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int len, ret, ch;
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int lame_result;
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if (frame) {
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switch (avctx->sample_fmt) {
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case AV_SAMPLE_FMT_S16P:
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ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
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break;
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case AV_SAMPLE_FMT_S32P:
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ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
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break;
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case AV_SAMPLE_FMT_FLTP:
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if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
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av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
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return AVERROR(EINVAL);
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}
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for (ch = 0; ch < avctx->channels; ch++) {
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s->fdsp.vector_fmul_scalar(s->samples_flt[ch],
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(const float *)frame->data[ch],
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32768.0f,
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FFALIGN(frame->nb_samples, 8));
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}
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ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
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break;
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default:
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return AVERROR_BUG;
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}
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} else {
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lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
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s->buffer_size - s->buffer_index);
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}
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if (lame_result < 0) {
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if (lame_result == -1) {
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av_log(avctx, AV_LOG_ERROR,
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"lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
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s->buffer_index, s->buffer_size - s->buffer_index);
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}
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return -1;
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}
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s->buffer_index += lame_result;
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ret = realloc_buffer(s);
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if (ret < 0) {
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av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
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return ret;
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}
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/* add current frame to the queue */
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if (frame) {
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if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
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return ret;
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}
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/* Move 1 frame from the LAME buffer to the output packet, if available.
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We have to parse the first frame header in the output buffer to
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determine the frame size. */
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if (s->buffer_index < 4)
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return 0;
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if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
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av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
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return -1;
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}
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len = hdr.frame_size;
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av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
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s->buffer_index);
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if (len <= s->buffer_index) {
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if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
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return ret;
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memcpy(avpkt->data, s->buffer, len);
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s->buffer_index -= len;
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memmove(s->buffer, s->buffer + len, s->buffer_index);
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/* Get the next frame pts/duration */
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ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
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&avpkt->duration);
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avpkt->size = len;
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*got_packet_ptr = 1;
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}
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return 0;
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}
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#define OFFSET(x) offsetof(LAMEContext, x)
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#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{ "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
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{ "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
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{ NULL },
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};
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static const AVClass libmp3lame_class = {
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.class_name = "libmp3lame encoder",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static const AVCodecDefault libmp3lame_defaults[] = {
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{ "b", "0" },
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{ NULL },
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};
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static const int libmp3lame_sample_rates[] = {
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44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
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};
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AVCodec ff_libmp3lame_encoder = {
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.name = "libmp3lame",
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.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_MP3,
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.priv_data_size = sizeof(LAMEContext),
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.init = mp3lame_encode_init,
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.encode2 = mp3lame_encode_frame,
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.close = mp3lame_encode_close,
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.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_NONE },
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.supported_samplerates = libmp3lame_sample_rates,
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.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_STEREO,
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0 },
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.priv_class = &libmp3lame_class,
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.defaults = libmp3lame_defaults,
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};
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