[FFmpeg-cvslog] avformat/audiointerleave: only keep the retime functionality of the audio interleaver
Marton Balint
git at videolan.org
Fri May 8 00:26:37 EEST 2020
ffmpeg | branch: master | Marton Balint <cus at passwd.hu> | Wed Mar 25 23:49:17 2020 +0100| [c5324d92c5f206dcdc2cf93ae237eaa7c1ad0a40] | committer: Marton Balint
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 <cus at passwd.hu>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c5324d92c5f206dcdc2cf93ae237eaa7c1ad0a40
---
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(-)
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 <baptiste dot coudurier at gmail dot com>
- *
- * 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 <baptiste dot coudurier at gmail dot com>
+ *
+ * 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 *));
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