diff options
Diffstat (limited to 'libavcodec/libmp3lame.c')
| -rw-r--r-- | libavcodec/libmp3lame.c | 112 |
1 files changed, 91 insertions, 21 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index 3ac033f758..d75183e9c0 100644 --- a/libavcodec/libmp3lame.c +++ b/libavcodec/libmp3lame.c @@ -2,20 +2,20 @@ * Interface to libmp3lame for mp3 encoding * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org> * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -31,13 +31,17 @@ #include "mpegaudio.h" #include <lame/lame.h> -#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4) +#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; @@ -45,8 +49,11 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) { Mp3AudioContext *s = avctx->priv_data; - if (avctx->channels > 2) - return -1; + 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; @@ -73,8 +80,25 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) goto err_close; avctx->frame_size = lame_get_framesize(s->gfp); - avctx->coded_frame = avcodec_alloc_frame(); - avctx->coded_frame->key_frame = 1; + + if(!(avctx->coded_frame= avcodec_alloc_frame())) { + lame_close(s->gfp); + + return AVERROR(ENOMEM); + } + + 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; @@ -152,21 +176,63 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, /* lame 3.91 dies on '1-channel interleaved' data */ - if (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) { - lame_result = lame_encode_buffer_interleaved(s->gfp, data, - avctx->frame_size, - s->buffer + s->buffer_index, - BUFFER_SIZE - s->buffer_index); + 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(s->gfp, data, data, - avctx->frame_size, s->buffer + - s->buffer_index, BUFFER_SIZE - - s->buffer_index); + 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 + ); } - } else { - lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index, - BUFFER_SIZE - s->buffer_index); } if (lame_result < 0) { @@ -205,6 +271,7 @@ 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); @@ -235,6 +302,9 @@ AVCodec ff_libmp3lame_encoder = { .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)"), |