| /* |
| * Interface to libmp3lame for mp3 encoding |
| * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org> |
| * |
| * 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 2.1 of the License, or (at your option) any later version. |
| * |
| * 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 FFmpeg; 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 "libavutil/intreadwrite.h" |
| #include "libavutil/log.h" |
| #include "libavutil/opt.h" |
| #include "avcodec.h" |
| #include "mpegaudio.h" |
| #include <lame/lame.h> |
| |
| #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. |
| typedef struct Mp3AudioContext { |
| AVClass *class; |
| lame_global_flags *gfp; |
| int stereo; |
| uint8_t buffer[BUFFER_SIZE]; |
| int buffer_index; |
| struct { |
| int *left; |
| int *right; |
| } s32_data; |
| int reservoir; |
| } Mp3AudioContext; |
| |
| static av_cold int MP3lame_encode_init(AVCodecContext *avctx) |
| { |
| Mp3AudioContext *s = avctx->priv_data; |
| |
| if (avctx->channels > 2) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Invalid number of channels %d, must be <= 2\n", avctx->channels); |
| return AVERROR(EINVAL); |
| } |
| |
| 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); |
| 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_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); |
| |
| if(!(avctx->coded_frame= avcodec_alloc_frame())) { |
| lame_close(s->gfp); |
| |
| return AVERROR(ENOMEM); |
| } |
| avctx->coded_frame->key_frame = 1; |
| |
| if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) { |
| int nelem = 2 * avctx->frame_size; |
| |
| if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) { |
| av_freep(&avctx->coded_frame); |
| lame_close(s->gfp); |
| |
| return AVERROR(ENOMEM); |
| } |
| |
| s->s32_data.right = s->s32_data.left + avctx->frame_size; |
| } |
| |
| return 0; |
| |
| err_close: |
| lame_close(s->gfp); |
| err: |
| return -1; |
| } |
| |
| 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, 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 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); |
| 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 temp0, temp1, bitRate; |
| |
| if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 || |
| sampleRateID == 3) { |
| return -1; |
| } |
| |
| if (!samplesPerFrame) |
| samplesPerFrame = &temp0; |
| if (!sampleRate) |
| sampleRate = &temp1; |
| |
| //*isMono = ((header >> 6) & 0x03) == 0x03; |
| |
| *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); |
| |
| return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; |
| } |
| |
| static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, |
| int buf_size, void *data) |
| { |
| Mp3AudioContext *s = avctx->priv_data; |
| int len; |
| int lame_result; |
| |
| /* lame 3.91 dies on '1-channel interleaved' data */ |
| |
| if (!data){ |
| lame_result= lame_encode_flush( |
| s->gfp, |
| s->buffer + s->buffer_index, |
| BUFFER_SIZE - s->buffer_index |
| ); |
| #if 2147483647 == INT_MAX |
| }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){ |
| if (s->stereo) { |
| int32_t *rp = data; |
| int32_t *mp = rp + 2*avctx->frame_size; |
| int *wpl = s->s32_data.left; |
| int *wpr = s->s32_data.right; |
| |
| while (rp < mp) { |
| *wpl++ = *rp++; |
| *wpr++ = *rp++; |
| } |
| |
| lame_result = lame_encode_buffer_int( |
| s->gfp, |
| s->s32_data.left, |
| s->s32_data.right, |
| avctx->frame_size, |
| s->buffer + s->buffer_index, |
| BUFFER_SIZE - s->buffer_index |
| ); |
| } else { |
| lame_result = lame_encode_buffer_int( |
| s->gfp, |
| data, |
| data, |
| avctx->frame_size, |
| s->buffer + s->buffer_index, |
| BUFFER_SIZE - s->buffer_index |
| ); |
| } |
| #endif |
| }else{ |
| 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 |
| ); |
| } else { |
| lame_result = lame_encode_buffer( |
| s->gfp, |
| data, |
| data, |
| avctx->frame_size, |
| s->buffer + s->buffer_index, |
| BUFFER_SIZE - s->buffer_index |
| ); |
| } |
| } |
| |
| 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) |
| 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 av_cold int MP3lame_encode_close(AVCodecContext *avctx) |
| { |
| Mp3AudioContext *s = avctx->priv_data; |
| |
| av_freep(&s->s32_data.left); |
| av_freep(&avctx->coded_frame); |
| |
| lame_close(s->gfp); |
| 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 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, |
| #if 2147483647 == INT_MAX |
| AV_SAMPLE_FMT_S32, |
| #endif |
| AV_SAMPLE_FMT_NONE }, |
| .supported_samplerates = sSampleRates, |
| .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), |
| .priv_class = &libmp3lame_class, |
| }; |