mirror of
https://git.videolan.org/git/ffmpeg.git
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243 lines
8.0 KiB
C
243 lines
8.0 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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 "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 "mpegaudio.h"
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#include <lame/lame.h>
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#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
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typedef struct Mp3AudioContext {
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AVClass *class;
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lame_global_flags *gfp;
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int stereo;
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uint8_t buffer[BUFFER_SIZE];
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int buffer_index;
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int reservoir;
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} Mp3AudioContext;
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static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
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{
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Mp3AudioContext *s = avctx->priv_data;
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if (avctx->channels > 2)
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return -1;
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s->stereo = avctx->channels > 1 ? 1 : 0;
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if ((s->gfp = lame_init()) == NULL)
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goto err;
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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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lame_set_num_channels(s->gfp, avctx->channels);
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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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}
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lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
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lame_set_brate(s->gfp, avctx->bit_rate / 1000);
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if (avctx->flags & CODEC_FLAG_QSCALE) {
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lame_set_brate(s->gfp, 0);
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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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}
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lame_set_bWriteVbrTag(s->gfp,0);
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lame_set_disable_reservoir(s->gfp, !s->reservoir);
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if (lame_init_params(s->gfp) < 0)
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goto err_close;
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avctx->frame_size = lame_get_framesize(s->gfp);
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avctx->coded_frame = avcodec_alloc_frame();
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avctx->coded_frame->key_frame = 1;
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return 0;
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err_close:
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lame_close(s->gfp);
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err:
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return -1;
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}
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static const int sSampleRates[] = {
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44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
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};
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static const int sBitRates[2][3][15] = {
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{
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{ 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
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{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
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{ 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
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},
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{
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{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
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},
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};
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static const int sSamplesPerFrame[2][3] = {
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{ 384, 1152, 1152 },
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{ 384, 1152, 576 }
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};
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static const int sBitsPerSlot[3] = { 32, 8, 8 };
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static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
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{
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uint32_t header = AV_RB32(data);
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int layerID = 3 - ((header >> 17) & 0x03);
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int bitRateID = ((header >> 12) & 0x0f);
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int sampleRateID = ((header >> 10) & 0x03);
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int bitsPerSlot = sBitsPerSlot[layerID];
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int isPadded = ((header >> 9) & 0x01);
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static int const mode_tab[4] = { 2, 3, 1, 0 };
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int mode = mode_tab[(header >> 19) & 0x03];
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int mpeg_id = mode > 0;
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int temp0, temp1, bitRate;
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if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
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sampleRateID == 3) {
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return -1;
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}
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if (!samplesPerFrame)
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samplesPerFrame = &temp0;
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if (!sampleRate)
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sampleRate = &temp1;
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//*isMono = ((header >> 6) & 0x03) == 0x03;
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*sampleRate = sSampleRates[sampleRateID] >> mode;
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bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
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*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
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//av_log(NULL, AV_LOG_DEBUG,
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// "sr:%d br:%d spf:%d l:%d m:%d\n",
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// *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
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return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
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}
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static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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int buf_size, void *data)
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{
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Mp3AudioContext *s = avctx->priv_data;
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int len;
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int lame_result;
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/* lame 3.91 dies on '1-channel interleaved' data */
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if (data) {
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if (s->stereo) {
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lame_result = lame_encode_buffer_interleaved(s->gfp, data,
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avctx->frame_size,
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s->buffer + s->buffer_index,
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BUFFER_SIZE - s->buffer_index);
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} else {
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lame_result = lame_encode_buffer(s->gfp, data, data,
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avctx->frame_size, s->buffer +
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s->buffer_index, BUFFER_SIZE -
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s->buffer_index);
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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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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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/* output buffer too small */
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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, 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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if (s->buffer_index < 4)
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return 0;
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len = mp3len(s->buffer, NULL, NULL);
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//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
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// avctx->frame_size, len, s->buffer_index);
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if (len <= s->buffer_index) {
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memcpy(frame, 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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// FIXME fix the audio codec API, so we do not need the memcpy()
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/*for(i=0; i<len; i++) {
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av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
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}*/
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return len;
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} else
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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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Mp3AudioContext *s = avctx->priv_data;
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av_freep(&avctx->coded_frame);
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lame_close(s->gfp);
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return 0;
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}
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#define OFFSET(x) offsetof(Mp3AudioContext, 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, { 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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AVCodec ff_libmp3lame_encoder = {
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.name = "libmp3lame",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_MP3,
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.priv_data_size = sizeof(Mp3AudioContext),
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.init = MP3lame_encode_init,
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.encode = MP3lame_encode_frame,
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.close = MP3lame_encode_close,
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.capabilities = CODEC_CAP_DELAY,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.supported_samplerates = sSampleRates,
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.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
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.priv_class = &libmp3lame_class,
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};
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