diff --git a/libavformat/Makefile b/libavformat/Makefile index e16a5ea8c2..5694bfc76c 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -116,7 +116,7 @@ OBJS-$(CONFIG_MSNWC_TCP_DEMUXER) += msnwc_tcp.o OBJS-$(CONFIG_MTV_DEMUXER) += mtv.o OBJS-$(CONFIG_MVI_DEMUXER) += mvi.o OBJS-$(CONFIG_MXF_DEMUXER) += mxfdec.o mxf.o -OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o +OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o audiointerleave.o OBJS-$(CONFIG_NSV_DEMUXER) += nsvdec.o OBJS-$(CONFIG_NULL_MUXER) += raw.o OBJS-$(CONFIG_NUT_DEMUXER) += nutdec.o nut.o riff.o diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c new file mode 100644 index 0000000000..e34026c408 --- /dev/null +++ b/libavformat/audiointerleave.c @@ -0,0 +1,125 @@ +/* + * Audio Interleaving functions + * + * Copyright (c) 2009 Baptiste Coudurier + * + * 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 + */ + +#include "libavutil/fifo.h" +#include "avformat.h" +#include "audiointerleave.h" + +void ff_audio_interleave_close(AVFormatContext *s) +{ + int i; + for (i = 0; i < s->nb_streams; i++) { + AVStream *st = s->streams[i]; + AudioInterleaveContext *aic = st->priv_data; + + if (st->codec->codec_type == CODEC_TYPE_AUDIO) + av_fifo_free(&aic->fifo); + } +} + +int ff_audio_interleave_init(AVFormatContext *s, + const int *samples_per_frame, + AVRational time_base) +{ + int i; + + if (!samples_per_frame) + return -1; + + for (i = 0; i < s->nb_streams; i++) { + AVStream *st = s->streams[i]; + AudioInterleaveContext *aic = st->priv_data; + + if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + aic->sample_size = (st->codec->channels * + av_get_bits_per_sample(st->codec->codec_id)) / 8; + if (!aic->sample_size) { + av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); + return -1; + } + aic->samples_per_frame = samples_per_frame; + aic->samples = aic->samples_per_frame; + aic->time_base = time_base; + + av_fifo_init(&aic->fifo, 100 * *aic->samples); + } + } + + return 0; +} + +int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, + int stream_index, int flush) +{ + AVStream *st = s->streams[stream_index]; + AudioInterleaveContext *aic = st->priv_data; + + int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size); + if (!size || (!flush && size == av_fifo_size(&aic->fifo))) + return 0; + + av_new_packet(pkt, size); + av_fifo_read(&aic->fifo, pkt->data, size); + + pkt->dts = pkt->pts = aic->dts; + pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); + pkt->stream_index = stream_index; + aic->dts += pkt->duration; + + aic->samples++; + if (!*aic->samples) + aic->samples = aic->samples_per_frame; + + return size; +} + +int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, + int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), + int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)) +{ + int i; + + if (pkt) { + AVStream *st = s->streams[pkt->stream_index]; + AudioInterleaveContext *aic = st->priv_data; + if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL); + } else { + // rewrite pts and dts to be decoded time line position + pkt->dts = aic->dts; + aic->dts += pkt->duration; + ff_interleave_add_packet(s, pkt, compare_ts); + } + pkt = NULL; + } + + for (i = 0; i < s->nb_streams; i++) { + AVStream *st = s->streams[i]; + if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + AVPacket new_pkt; + while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) + ff_interleave_add_packet(s, &new_pkt, compare_ts); + } + } + + return get_packet(s, out, pkt, flush); +} diff --git a/libavformat/audiointerleave.h b/libavformat/audiointerleave.h new file mode 100644 index 0000000000..a50e7a61bf --- /dev/null +++ b/libavformat/audiointerleave.h @@ -0,0 +1,49 @@ +/* + * Audio Interleaving prototypes and declarations + * + * Copyright (c) 2009 Baptiste Coudurier + * + * 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 + */ + +#ifndef AVFORMAT_AUDIOINTERLEAVE_H +#define AVFORMAT_AUDIOINTERLEAVE_H + +#include "libavutil/fifo.h" +#include "avformat.h" + +typedef struct { + AVFifoBuffer fifo; + unsigned fifo_size; ///< current fifo size allocated + uint64_t dts; ///< current dts + int sample_size; ///< size of one sample all channels included + const int *samples_per_frame; ///< must be 0 terminated + const int *samples; ///< current samples per frame, pointer to samples_per_frame + AVRational time_base; ///< time base of output audio packets +} AudioInterleaveContext; + +int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base); +void ff_audio_interleave_close(AVFormatContext *s); + +int ff_interleave_compare_dts(AVFormatContext *s, AVPacket *next, AVPacket *pkt); +int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, + int stream_index, int flush); +int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, + int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), + int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)); + +#endif // AVFORMAT_AUDIOINTERLEAVE_H diff --git a/libavformat/mxfenc.c b/libavformat/mxfenc.c index 65f18bc395..8630403423 100644 --- a/libavformat/mxfenc.c +++ b/libavformat/mxfenc.c @@ -36,6 +36,7 @@ #include #include "libavutil/fifo.h" +#include "audiointerleave.h" #include "mxf.h" static const int NTSC_samples_per_frame[] = { 1602, 1601, 1602, 1601, 1602, 0 }; @@ -44,16 +45,6 @@ static const int PAL_samples_per_frame[] = { 1920, 0 }; #define MXF_INDEX_CLUSTER_SIZE 4096 #define KAG_SIZE 512 -typedef struct { - AVFifoBuffer fifo; - unsigned fifo_size; ///< current fifo size allocated - uint64_t dts; ///< current dts - int sample_size; ///< size of one sample all channels included - const int *samples_per_frame; ///< must be 0 terminated - const int *samples; ///< current samples per frame, pointer to samples_per_frame - AVRational time_base; ///< time base of output audio packets -} AudioInterleaveContext; - typedef struct { int local_tag; UID uid; @@ -1110,49 +1101,6 @@ static int mxf_parse_mpeg2_frame(AVFormatContext *s, AVStream *st, AVPacket *pkt return !!sc->codec_ul; } -static int ff_audio_interleave_init(AVFormatContext *s, - const int *samples_per_frame, - AVRational time_base) -{ - int i; - - if (!samples_per_frame) - return -1; - - for (i = 0; i < s->nb_streams; i++) { - AVStream *st = s->streams[i]; - AudioInterleaveContext *aic = st->priv_data; - - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { - aic->sample_size = (st->codec->channels * - av_get_bits_per_sample(st->codec->codec_id)) / 8; - if (!aic->sample_size) { - av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); - return -1; - } - aic->samples_per_frame = samples_per_frame; - aic->samples = aic->samples_per_frame; - aic->time_base = time_base; - - av_fifo_init(&aic->fifo, 100 * *aic->samples); - } - } - - return 0; -} - -static void ff_audio_interleave_close(AVFormatContext *s) -{ - int i; - for (i = 0; i < s->nb_streams; i++) { - AVStream *st = s->streams[i]; - AudioInterleaveContext *aic = st->priv_data; - - if (st->codec->codec_type == CODEC_TYPE_AUDIO) - av_fifo_free(&aic->fifo); - } -} - static uint64_t mxf_parse_timestamp(time_t timestamp) { struct tm *time = localtime(×tamp); @@ -1428,31 +1376,6 @@ static int mxf_write_footer(AVFormatContext *s) return 0; } -static int mxf_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, - int stream_index, int flush) -{ - AVStream *st = s->streams[stream_index]; - AudioInterleaveContext *aic = st->priv_data; - - int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size); - if (!size || (!flush && size == av_fifo_size(&aic->fifo))) - return 0; - - av_new_packet(pkt, size); - av_fifo_read(&aic->fifo, pkt->data, size); - - pkt->dts = pkt->pts = aic->dts; - pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); - pkt->stream_index = stream_index; - aic->dts += pkt->duration; - - aic->samples++; - if (!*aic->samples) - aic->samples = aic->samples_per_frame; - - return size; -} - static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush) { AVPacketList *pktl; @@ -1517,32 +1440,8 @@ static int mxf_compare_timestamps(AVFormatContext *s, AVPacket *next, AVPacket * static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush) { - int i; - - if (pkt) { - AVStream *st = s->streams[pkt->stream_index]; - AudioInterleaveContext *aic = st->priv_data; - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { - av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL); - } else { - // rewrite pts and dts to be decoded time line position - pkt->pts = pkt->dts = aic->dts; - aic->dts += pkt->duration; - ff_interleave_add_packet(s, pkt, mxf_compare_timestamps); - } - pkt = NULL; - } - - for (i = 0; i < s->nb_streams; i++) { - AVStream *st = s->streams[i]; - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { - AVPacket new_pkt; - while (mxf_interleave_new_audio_packet(s, &new_pkt, i, flush)) - ff_interleave_add_packet(s, &new_pkt, mxf_compare_timestamps); - } - } - - return mxf_interleave_get_packet(s, out, pkt, flush); + return ff_audio_interleave(s, out, pkt, flush, + mxf_interleave_get_packet, mxf_compare_timestamps); } AVOutputFormat mxf_muxer = {