diff options
Diffstat (limited to 'libavcodec/libmp3lame.c')
| -rw-r--r-- | libavcodec/libmp3lame.c | 41 |
1 files changed, 23 insertions, 18 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index 067a17f..b5d5004 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 */ @@ -38,7 +38,7 @@ #include "mpegaudio.h" #include "mpegaudiodecheader.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 LAMEContext { AVClass *class; @@ -52,7 +52,7 @@ typedef struct LAMEContext { int abr; float *samples_flt[2]; AudioFrameQueue afq; - AVFloatDSPContext fdsp; + AVFloatDSPContext *fdsp; } LAMEContext; @@ -79,6 +79,7 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx) av_freep(&s->samples_flt[0]); av_freep(&s->samples_flt[1]); av_freep(&s->buffer); + av_freep(&s->fdsp); ff_af_queue_close(&s->afq); @@ -97,6 +98,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx) if (!(s->gfp = lame_init())) return AVERROR(ENOMEM); + lame_set_num_channels(s->gfp, avctx->channels); lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO); @@ -105,9 +107,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx) lame_set_out_samplerate(s->gfp, avctx->sample_rate); /* algorithmic quality */ - if (avctx->compression_level == FF_COMPRESSION_DEFAULT) - lame_set_quality(s->gfp, 5); - else + if (avctx->compression_level != FF_COMPRESSION_DEFAULT) lame_set_quality(s->gfp, avctx->compression_level); /* rate control */ @@ -146,7 +146,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx) if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) { int ch; for (ch = 0; ch < avctx->channels; ch++) { - s->samples_flt[ch] = av_malloc(avctx->frame_size * + s->samples_flt[ch] = av_malloc_array(avctx->frame_size, sizeof(*s->samples_flt[ch])); if (!s->samples_flt[ch]) { ret = AVERROR(ENOMEM); @@ -159,7 +159,12 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx) if (ret < 0) goto error; - avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); + s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); + if (!s->fdsp) { + ret = AVERROR(ENOMEM); + goto error; + } + return 0; error: @@ -198,7 +203,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, return AVERROR(EINVAL); } for (ch = 0; ch < avctx->channels; ch++) { - s->fdsp.vector_fmul_scalar(s->samples_flt[ch], + s->fdsp->vector_fmul_scalar(s->samples_flt[ch], (const float *)frame->data[ch], 32768.0f, FFALIGN(frame->nb_samples, 8)); @@ -208,6 +213,8 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, default: return AVERROR_BUG; } + } else if (!s->afq.frame_alloc) { + lame_result = 0; } else { lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index, s->buffer_size - s->buffer_index); @@ -251,10 +258,8 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index); if (len <= s->buffer_index) { - if ((ret = ff_alloc_packet(avpkt, len))) { - av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0) return ret; - } memcpy(avpkt->data, s->buffer, len); s->buffer_index -= len; memmove(s->buffer, s->buffer + len, s->buffer_index); @@ -272,9 +277,9 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, #define OFFSET(x) offsetof(LAMEContext, 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, { .i64 = 1 }, 0, 1, AE }, - { "joint_stereo", "Use joint stereo.", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE }, - { "abr", "Use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE }, + { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE }, + { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE }, + { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE }, { NULL }, }; |
