From c5324d92c5f206dcdc2cf93ae237eaa7c1ad0a40 Mon Sep 17 00:00:00 2001 From: Marton Balint Date: Wed, 25 Mar 2020 23:49:17 +0100 Subject: [PATCH] avformat/audiointerleave: only keep the retime functionality of the audio interleaver And rename it to retimeinterleave, use the pcm_rechunk bitstream filter for rechunking. By seperating the two functions we hopefully get cleaner code. Signed-off-by: Marton Balint --- configure | 2 + libavformat/Makefile | 4 +- libavformat/audiointerleave.c | 148 ------------------ libavformat/gxfenc.c | 20 ++- libavformat/mxfenc.c | 19 +-- libavformat/retimeinterleave.c | 51 ++++++ .../{audiointerleave.h => retimeinterleave.h} | 31 ++-- 7 files changed, 87 insertions(+), 188 deletions(-) delete mode 100644 libavformat/audiointerleave.c create mode 100644 libavformat/retimeinterleave.c rename libavformat/{audiointerleave.h => retimeinterleave.h} (57%) diff --git a/configure b/configure index 080d93a129..e7162dbc56 100755 --- a/configure +++ b/configure @@ -2722,6 +2722,7 @@ fraps_decoder_select="bswapdsp huffman" g2m_decoder_deps="zlib" g2m_decoder_select="blockdsp idctdsp jpegtables" g729_decoder_select="audiodsp" +gxf_encoder_select="pcm_rechunk_bsf" h261_decoder_select="mpegvideo" h261_encoder_select="mpegvideoenc" h263_decoder_select="h263_parser h263dsp mpegvideo qpeldsp" @@ -2794,6 +2795,7 @@ mv30_decoder_select="aandcttables blockdsp" mvha_decoder_deps="zlib" mvha_decoder_select="llviddsp" mwsc_decoder_deps="zlib" +mxf_encoder_select="pcm_rechunk_bsf" mxpeg_decoder_select="mjpeg_decoder" nellymoser_decoder_select="mdct sinewin" nellymoser_encoder_select="audio_frame_queue mdct sinewin" diff --git a/libavformat/Makefile b/libavformat/Makefile index b744eb69b2..ec01c6c65c 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -205,7 +205,7 @@ OBJS-$(CONFIG_GIF_DEMUXER) += gifdec.o OBJS-$(CONFIG_GSM_DEMUXER) += gsmdec.o OBJS-$(CONFIG_GSM_MUXER) += rawenc.o OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o -OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o +OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o retimeinterleave.o OBJS-$(CONFIG_G722_DEMUXER) += g722.o rawdec.o OBJS-$(CONFIG_G722_MUXER) += rawenc.o OBJS-$(CONFIG_G723_1_DEMUXER) += g723_1.o @@ -347,7 +347,7 @@ OBJS-$(CONFIG_MUSX_DEMUXER) += musx.o OBJS-$(CONFIG_MV_DEMUXER) += mvdec.o OBJS-$(CONFIG_MVI_DEMUXER) += mvi.o OBJS-$(CONFIG_MXF_DEMUXER) += mxfdec.o mxf.o -OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o audiointerleave.o avc.o +OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o retimeinterleave.o avc.o OBJS-$(CONFIG_MXG_DEMUXER) += mxg.o OBJS-$(CONFIG_NC_DEMUXER) += ncdec.o OBJS-$(CONFIG_NISTSPHERE_DEMUXER) += nistspheredec.o pcm.o diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c deleted file mode 100644 index 36a3288242..0000000000 --- a/libavformat/audiointerleave.c +++ /dev/null @@ -1,148 +0,0 @@ -/* - * 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 "libavutil/mathematics.h" -#include "avformat.h" -#include "audiointerleave.h" -#include "internal.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 (aic && st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) - av_fifo_freep(&aic->fifo); - } -} - -int ff_audio_interleave_init(AVFormatContext *s, - const int samples_per_frame, - AVRational time_base) -{ - int i; - - if (!time_base.num) { - av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); - return AVERROR(EINVAL); - } - for (i = 0; i < s->nb_streams; i++) { - AVStream *st = s->streams[i]; - AudioInterleaveContext *aic = st->priv_data; - - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - int max_samples = samples_per_frame ? samples_per_frame : - av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP); - aic->sample_size = (st->codecpar->channels * - av_get_bits_per_sample(st->codecpar->codec_id)) / 8; - if (!aic->sample_size) { - av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); - return AVERROR(EINVAL); - } - aic->samples_per_frame = samples_per_frame; - aic->time_base = time_base; - - if (!(aic->fifo = av_fifo_alloc_array(100, max_samples))) - return AVERROR(ENOMEM); - aic->fifo_size = 100 * max_samples; - } - } - - return 0; -} - -static int 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 ret; - int nb_samples = aic->samples_per_frame ? aic->samples_per_frame : - (av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples); - int frame_size = nb_samples * aic->sample_size; - int size = FFMIN(av_fifo_size(aic->fifo), frame_size); - if (!size || (!flush && size == av_fifo_size(aic->fifo))) - return 0; - - ret = av_new_packet(pkt, frame_size); - if (ret < 0) - return ret; - av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); - - if (size < pkt->size) - memset(pkt->data + size, 0, pkt->size - size); - - pkt->dts = pkt->pts = aic->dts; - pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base); - pkt->stream_index = stream_index; - aic->dts += pkt->duration; - aic->nb_samples += nb_samples; - aic->n++; - - return pkt->size; -} - -int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, - int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), - int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *)) -{ - int i, ret; - - if (pkt) { - AVStream *st = s->streams[pkt->stream_index]; - AudioInterleaveContext *aic = st->priv_data; - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; - if (new_size > aic->fifo_size) { - if (av_fifo_realloc2(aic->fifo, new_size) < 0) - return AVERROR(ENOMEM); - aic->fifo_size = new_size; - } - 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; - if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0) - return ret; - } - pkt = NULL; - } - - for (i = 0; i < s->nb_streams; i++) { - AVStream *st = s->streams[i]; - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - AVPacket new_pkt; - while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { - if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0) - return ret; - } - if (ret < 0) - return ret; - } - } - - return get_packet(s, out, NULL, flush); -} diff --git a/libavformat/gxfenc.c b/libavformat/gxfenc.c index e7536a6a7e..60468c36ce 100644 --- a/libavformat/gxfenc.c +++ b/libavformat/gxfenc.c @@ -27,8 +27,9 @@ #include "avformat.h" #include "internal.h" #include "gxf.h" -#include "audiointerleave.h" +#include "retimeinterleave.h" +#define GXF_SAMPLES_PER_FRAME 32768 #define GXF_AUDIO_PACKET_SIZE 65536 #define GXF_TIMECODE(c, d, h, m, s, f) \ @@ -44,7 +45,7 @@ typedef struct GXFTimecode{ } GXFTimecode; typedef struct GXFStreamContext { - AudioInterleaveContext aic; + RetimeInterleaveContext aic; uint32_t track_type; uint32_t sample_size; uint32_t sample_rate; @@ -663,8 +664,6 @@ static int gxf_write_umf_packet(AVFormatContext *s) return updatePacketSize(pb, pos); } -static const int GXF_samples_per_frame = 32768; - static void gxf_init_timecode_track(GXFStreamContext *sc, GXFStreamContext *vsc) { if (!vsc) @@ -736,6 +735,9 @@ static int gxf_write_header(AVFormatContext *s) av_log(s, AV_LOG_ERROR, "only mono tracks are allowed\n"); return -1; } + ret = ff_stream_add_bitstream_filter(st, "pcm_rechunk", "n="AV_STRINGIFY(GXF_SAMPLES_PER_FRAME)); + if (ret < 0) + return ret; sc->track_type = 2; sc->sample_rate = st->codecpar->sample_rate; avpriv_set_pts_info(st, 64, 1, sc->sample_rate); @@ -813,14 +815,12 @@ static int gxf_write_header(AVFormatContext *s) return -1; } } + ff_retime_interleave_init(&sc->aic, st->time_base); /* FIXME first 10 audio tracks are 0 to 9 next 22 are A to V */ sc->media_info = media_info<<8 | ('0'+tracks[media_info]++); sc->order = s->nb_streams - st->index; } - if (ff_audio_interleave_init(s, GXF_samples_per_frame, (AVRational){ 1, 48000 }) < 0) - return -1; - if (tcr && vsc) gxf_init_timecode(s, &gxf->tc, tcr->value, vsc->fields); @@ -877,8 +877,6 @@ static void gxf_deinit(AVFormatContext *s) { GXFContext *gxf = s->priv_data; - ff_audio_interleave_close(s); - av_freep(&gxf->flt_entries); av_freep(&gxf->map_offsets); } @@ -1016,8 +1014,8 @@ static int gxf_interleave_packet(AVFormatContext *s, AVPacket *out, AVPacket *pk { if (pkt && s->streams[pkt->stream_index]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) pkt->duration = 2; // enforce 2 fields - return ff_audio_rechunk_interleave(s, out, pkt, flush, - ff_interleave_packet_per_dts, gxf_compare_field_nb); + return ff_retime_interleave(s, out, pkt, flush, + ff_interleave_packet_per_dts, gxf_compare_field_nb); } AVOutputFormat ff_gxf_muxer = { diff --git a/libavformat/mxfenc.c b/libavformat/mxfenc.c index 23147e9b84..63a2799b08 100644 --- a/libavformat/mxfenc.c +++ b/libavformat/mxfenc.c @@ -52,7 +52,7 @@ #include "libavcodec/h264_ps.h" #include "libavcodec/golomb.h" #include "libavcodec/internal.h" -#include "audiointerleave.h" +#include "retimeinterleave.h" #include "avformat.h" #include "avio_internal.h" #include "internal.h" @@ -79,7 +79,7 @@ typedef struct MXFIndexEntry { } MXFIndexEntry; typedef struct MXFStreamContext { - AudioInterleaveContext aic; + RetimeInterleaveContext aic; UID track_essence_element_key; int index; ///< index in mxf_essence_container_uls table const UID *codec_ul; @@ -2538,6 +2538,7 @@ static int mxf_write_header(AVFormatContext *s) if (mxf->signal_standard >= 0) sc->signal_standard = mxf->signal_standard; } else if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { + char bsf_arg[32]; if (st->codecpar->sample_rate != 48000) { av_log(s, AV_LOG_ERROR, "only 48khz is implemented\n"); return -1; @@ -2580,6 +2581,10 @@ static int mxf_write_header(AVFormatContext *s) av_rescale_rnd(st->codecpar->sample_rate, mxf->time_base.num, mxf->time_base.den, AV_ROUND_UP) * av_get_bits_per_sample(st->codecpar->codec_id) / 8; } + snprintf(bsf_arg, sizeof(bsf_arg), "r=%d/%d", mxf->tc.rate.num, mxf->tc.rate.den); + ret = ff_stream_add_bitstream_filter(st, "pcm_rechunk", bsf_arg); + if (ret < 0) + return ret; } else if (st->codecpar->codec_type == AVMEDIA_TYPE_DATA) { AVDictionaryEntry *e = av_dict_get(st->metadata, "data_type", NULL, 0); if (e && !strcmp(e->value, "vbi_vanc_smpte_436M")) { @@ -2593,6 +2598,7 @@ static int mxf_write_header(AVFormatContext *s) return -1; } } + ff_retime_interleave_init(&sc->aic, av_inv_q(mxf->tc.rate)); if (sc->index == -1) { sc->index = mxf_get_essence_container_ul_index(st->codecpar->codec_id); @@ -2646,9 +2652,6 @@ static int mxf_write_header(AVFormatContext *s) return AVERROR(ENOMEM); mxf->timecode_track->index = -1; - if (ff_audio_interleave_init(s, 0, av_inv_q(mxf->tc.rate)) < 0) - return -1; - return 0; } @@ -3010,8 +3013,6 @@ static void mxf_deinit(AVFormatContext *s) { MXFContext *mxf = s->priv_data; - ff_audio_interleave_close(s); - av_freep(&mxf->index_entries); av_freep(&mxf->body_partition_offset); if (mxf->timecode_track) { @@ -3086,8 +3087,8 @@ static int mxf_compare_timestamps(AVFormatContext *s, const AVPacket *next, static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush) { - return ff_audio_rechunk_interleave(s, out, pkt, flush, - mxf_interleave_get_packet, mxf_compare_timestamps); + return ff_retime_interleave(s, out, pkt, flush, + mxf_interleave_get_packet, mxf_compare_timestamps); } #define MXF_COMMON_OPTIONS \ diff --git a/libavformat/retimeinterleave.c b/libavformat/retimeinterleave.c new file mode 100644 index 0000000000..9f874e3626 --- /dev/null +++ b/libavformat/retimeinterleave.c @@ -0,0 +1,51 @@ +/* + * Retime 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/mathematics.h" +#include "avformat.h" +#include "retimeinterleave.h" +#include "internal.h" + +void ff_retime_interleave_init(RetimeInterleaveContext *aic, AVRational time_base) +{ + aic->time_base = time_base; +} + +int ff_retime_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, + int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), + int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *)) +{ + int ret; + + if (pkt) { + AVStream *st = s->streams[pkt->stream_index]; + RetimeInterleaveContext *aic = st->priv_data; + pkt->duration = av_rescale_q(pkt->duration, st->time_base, aic->time_base); + // rewrite pts and dts to be decoded time line position + pkt->pts = pkt->dts = aic->dts; + aic->dts += pkt->duration; + if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0) + return ret; + } + + return get_packet(s, out, NULL, flush); +} diff --git a/libavformat/audiointerleave.h b/libavformat/retimeinterleave.h similarity index 57% rename from libavformat/audiointerleave.h rename to libavformat/retimeinterleave.h index 0933310f4c..de0a7442b0 100644 --- a/libavformat/audiointerleave.h +++ b/libavformat/retimeinterleave.h @@ -20,36 +20,31 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVFORMAT_AUDIOINTERLEAVE_H -#define AVFORMAT_AUDIOINTERLEAVE_H +#ifndef AVFORMAT_RETIMEINTERLEAVE_H +#define AVFORMAT_RETIMEINTERLEAVE_H -#include "libavutil/fifo.h" #include "avformat.h" -typedef struct AudioInterleaveContext { - AVFifoBuffer *fifo; - unsigned fifo_size; ///< size of currently allocated FIFO - int64_t n; ///< number of generated packets - int64_t nb_samples; ///< number of generated samples +typedef struct RetimeInterleaveContext { uint64_t dts; ///< current dts - int sample_size; ///< size of one sample all channels included - int samples_per_frame; ///< samples per frame if fixed, 0 otherwise - AVRational time_base; ///< time base of output audio packets -} AudioInterleaveContext; + AVRational time_base; ///< time base of output packets +} RetimeInterleaveContext; -int ff_audio_interleave_init(AVFormatContext *s, const int samples_per_frame, AVRational time_base); -void ff_audio_interleave_close(AVFormatContext *s); +/** + * Init the retime interleave context + */ +void ff_retime_interleave_init(RetimeInterleaveContext *aic, AVRational time_base); /** - * Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame - * and interleave them correctly. - * The first element of AVStream->priv_data must be AudioInterleaveContext + * Retime packets per RetimeInterleaveContext->time_base and interleave them + * correctly. + * The first element of AVStream->priv_data must be RetimeInterleaveContext * when using this function. * * @param get_packet function will output a packet when streams are correctly interleaved. * @param compare_ts function will compare AVPackets and decide interleaving order. */ -int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, +int ff_retime_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *));