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@passwd.hu>

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"

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

libavformat/audiointerleave.c

-/*
- * 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);
-}

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 = {

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 \

libavformat/retimeinterleave.c

+/*
+ * 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);
+}

libavformat/audiointerleave.h → 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 *));