diff options
Diffstat (limited to 'libavcodec/libmp3lame.c')
| -rw-r--r-- | libavcodec/libmp3lame.c | 80 |
1 files changed, 69 insertions, 11 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index 6da6d71134..fcefa2915e 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 */ @@ -38,6 +38,10 @@ typedef struct Mp3AudioContext { int stereo; uint8_t buffer[BUFFER_SIZE]; int buffer_index; + struct { + int *left; + int *right; + } s32_data; int reservoir; } Mp3AudioContext; @@ -77,9 +81,26 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) avctx->frame_size = lame_get_framesize(s->gfp); - avctx->coded_frame= avcodec_alloc_frame(); + 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: @@ -154,17 +175,35 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, /* 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( + 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, - data, + 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( + lame_result = lame_encode_buffer_int( s->gfp, data, data, @@ -173,12 +212,26 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, BUFFER_SIZE - s->buffer_index ); } +#endif }else{ - lame_result= lame_encode_flush( + 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){ @@ -214,6 +267,7 @@ 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); @@ -243,7 +297,11 @@ AVCodec ff_libmp3lame_encoder = { .encode = MP3lame_encode_frame, .close = MP3lame_encode_close, .capabilities= CODEC_CAP_DELAY, - .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, + .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, |