@@ -59,6 +59,7 @@ version <next>:
- mv30 decoder
- Expanded styling support for 3GPP Timed Text Subtitles (movtext)
- WebP parser
+- pcm_rechunk bitstream filter
version 4.2:
@@ -548,6 +548,36 @@ ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@section null
This bitstream filter passes the packets through unchanged.
+@section pcm_rechunk
+
+Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
+rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
+filter,ffmpeg-filters} but works on audio packets instead of audio frames.
+
+@table @option
+@item nb_out_samples, n
+Set the number of samples per each output audio packet. The number is intended
+as the number of samples @emph{per each channel}. Default value is 1024.
+
+@item pad, p
+If set to 1, the filter will pad the last audio packet with silence, so that it
+will contain the same number of samples (or roughly the same number of samples,
+see @option{frame_rate}) as the previous ones. Default value is 1.
+
+@item frame_rate, r
+This option makes the filter output a fixed numer of packets per second instead
+of a fixed number of samples per packet. If the audio sample rate is not
+divisible by the frame rate then the number of samples will not be constant but
+will vary slightly so that each packet will start as close as to the frame
+boundary as possible. Using this option has precedence over @option{nb_out_samples}.
+@end table
+
+You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
+for NTSC frame rate using the @option{frame_rate} option.
+@example
+ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
+@end example
+
@section prores_metadata
Modify color property metadata embedded in prores stream.
@@ -1115,6 +1115,7 @@ OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o \
OBJS-$(CONFIG_MPEG2_METADATA_BSF) += mpeg2_metadata_bsf.o
OBJS-$(CONFIG_NOISE_BSF) += noise_bsf.o
OBJS-$(CONFIG_NULL_BSF) += null_bsf.o
+OBJS-$(CONFIG_PCM_RECHUNK_BSF) += pcm_rechunk_bsf.o
OBJS-$(CONFIG_PRORES_METADATA_BSF) += prores_metadata_bsf.o
OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += remove_extradata_bsf.o
OBJS-$(CONFIG_TEXT2MOVSUB_BSF) += movsub_bsf.o
@@ -49,6 +49,7 @@ extern const AVBitStreamFilter ff_mpeg4_unpack_bframes_bsf;
extern const AVBitStreamFilter ff_mov2textsub_bsf;
extern const AVBitStreamFilter ff_noise_bsf;
extern const AVBitStreamFilter ff_null_bsf;
+extern const AVBitStreamFilter ff_pcm_rechunk_bsf;
extern const AVBitStreamFilter ff_prores_metadata_bsf;
extern const AVBitStreamFilter ff_remove_extradata_bsf;
extern const AVBitStreamFilter ff_text2movsub_bsf;
new file mode 100644
@@ -0,0 +1,204 @@
+/*
+ * Copyright (c) 2020 Marton Balint
+ *
+ * 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 "avcodec.h"
+#include "bsf.h"
+#include "libavutil/avassert.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+
+typedef struct PCMContext {
+ const AVClass *class;
+
+ int nb_out_samples;
+ int pad;
+ AVRational frame_rate;
+
+ AVPacket *in_pkt;
+ AVPacket *out_pkt;
+ int sample_size;
+ int64_t n;
+ int64_t dts;
+} PCMContext;
+
+static int init(AVBSFContext *ctx)
+{
+ PCMContext *s = ctx->priv_data;
+ AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
+ int64_t min_samples;
+
+ ctx->time_base_out = av_inv_q(sr);
+
+ s->sample_size = ctx->par_in->channels * av_get_bits_per_sample(ctx->par_in->codec_id) / 8;
+
+ if (s->frame_rate.num) {
+ min_samples = av_rescale_q_rnd(1, sr, s->frame_rate, AV_ROUND_DOWN);
+ } else {
+ min_samples = s->nb_out_samples;
+ }
+ if (min_samples <= 0 || min_samples > INT_MAX / s->sample_size)
+ return AVERROR(EINVAL);
+
+ s->in_pkt = av_packet_alloc();
+ s->out_pkt = av_packet_alloc();
+ if (!s->in_pkt || !s->out_pkt)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static void uninit(AVBSFContext *ctx)
+{
+ PCMContext *s = ctx->priv_data;
+ av_packet_free(&s->in_pkt);
+ av_packet_free(&s->out_pkt);
+}
+
+static void flush(AVBSFContext *ctx)
+{
+ PCMContext *s = ctx->priv_data;
+ av_packet_unref(s->in_pkt);
+ av_packet_unref(s->out_pkt);
+ s->n = 0;
+ s->dts = 0;
+}
+
+static int send_packet(PCMContext *s, int nb_samples, AVPacket *pkt)
+{
+ pkt->dts = pkt->pts = s->dts;
+ pkt->duration = nb_samples;
+ s->dts += nb_samples;
+ s->n++;
+ return 0;
+}
+
+static int rechunk_filter(AVBSFContext *ctx, AVPacket *pkt)
+{
+ PCMContext *s = ctx->priv_data;
+ AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
+ int nb_samples = s->frame_rate.num ? (av_rescale_q(s->n + 1, sr, s->frame_rate) - s->dts) : s->nb_out_samples;
+ int data_size = nb_samples * s->sample_size;
+ int ret;
+
+ do {
+ if (s->in_pkt->size) {
+ if (s->out_pkt->size || s->in_pkt->size < data_size) {
+ int drain = FFMIN(s->in_pkt->size, data_size - s->out_pkt->size);
+ if (!s->out_pkt->size) {
+ ret = av_new_packet(s->out_pkt, data_size);
+ if (ret < 0)
+ return ret;
+ ret = av_packet_copy_props(s->out_pkt, s->in_pkt);
+ if (ret < 0) {
+ av_packet_unref(s->out_pkt);
+ return ret;
+ }
+ s->out_pkt->size = 0;
+ }
+ memcpy(s->out_pkt->data + s->out_pkt->size, s->in_pkt->data, drain);
+ s->out_pkt->size += drain;
+ s->in_pkt->size -= drain;
+ s->in_pkt->data += drain;
+ if (s->out_pkt->size == data_size) {
+ av_packet_move_ref(pkt, s->out_pkt);
+ if (!s->in_pkt->size)
+ av_packet_unref(s->in_pkt);
+ return send_packet(s, nb_samples, pkt);
+ }
+ av_packet_unref(s->in_pkt);
+ } else if (s->in_pkt->size > data_size) {
+ ret = av_packet_ref(pkt, s->in_pkt);
+ if (ret < 0)
+ return ret;
+ pkt->size = data_size;
+ s->in_pkt->size -= data_size;
+ s->in_pkt->data += data_size;
+ return send_packet(s, nb_samples, pkt);
+ } else {
+ av_assert0(s->in_pkt->size == data_size);
+ av_packet_move_ref(pkt, s->in_pkt);
+ return send_packet(s, nb_samples, pkt);
+ }
+ }
+
+ ret = ff_bsf_get_packet_ref(ctx, s->in_pkt);
+ if (ret == AVERROR_EOF && s->out_pkt->size) {
+ if (s->pad) {
+ memset(s->out_pkt->data + s->out_pkt->size, 0, data_size - s->out_pkt->size);
+ s->out_pkt->size = data_size;
+ } else {
+ nb_samples = s->out_pkt->size / s->sample_size;
+ }
+ av_packet_move_ref(pkt, s->out_pkt);
+ return send_packet(s, nb_samples, pkt);
+ }
+ } while (ret >= 0);
+
+ return ret;
+}
+
+#define OFFSET(x) offsetof(PCMContext, x)
+#define FLAGS (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_BSF_PARAM)
+static const AVOption options[] = {
+ { "nb_out_samples", "set the number of per-packet output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
+ { "n", "set the number of per-packet output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
+ { "pad", "pad last packet with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1} , 0, 1, FLAGS },
+ { "p", "pad last packet with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1} , 0, 1, FLAGS },
+ { "frame_rate", "set number of packets per second", OFFSET(frame_rate), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, INT_MAX, FLAGS },
+ { "r", "set number of packets per second", OFFSET(frame_rate), AV_OPT_TYPE_RATIONAL, {.dbl=0}, 0, INT_MAX, FLAGS },
+ { NULL },
+};
+
+static const AVClass pcm_rechunk_class = {
+ .class_name = "pcm_rechunk_bsf",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+static const enum AVCodecID codec_ids[] = {
+ AV_CODEC_ID_PCM_S16LE,
+ AV_CODEC_ID_PCM_S16BE,
+ AV_CODEC_ID_PCM_S8,
+ AV_CODEC_ID_PCM_S32LE,
+ AV_CODEC_ID_PCM_S32BE,
+ AV_CODEC_ID_PCM_S24LE,
+ AV_CODEC_ID_PCM_S24BE,
+ AV_CODEC_ID_PCM_F32BE,
+ AV_CODEC_ID_PCM_F32LE,
+ AV_CODEC_ID_PCM_F64BE,
+ AV_CODEC_ID_PCM_F64LE,
+ AV_CODEC_ID_PCM_S64LE,
+ AV_CODEC_ID_PCM_S64BE,
+ AV_CODEC_ID_PCM_F16LE,
+ AV_CODEC_ID_PCM_F24LE,
+ AV_CODEC_ID_NONE,
+};
+
+const AVBitStreamFilter ff_pcm_rechunk_bsf = {
+ .name = "pcm_rechunk",
+ .priv_data_size = sizeof(PCMContext),
+ .priv_class = &pcm_rechunk_class,
+ .filter = rechunk_filter,
+ .init = init,
+ .flush = flush,
+ .close = uninit,
+ .codec_ids = codec_ids,
+};
@@ -28,7 +28,7 @@
#include "libavutil/version.h"
#define LIBAVCODEC_VERSION_MAJOR 58
-#define LIBAVCODEC_VERSION_MINOR 80
+#define LIBAVCODEC_VERSION_MINOR 81
#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
Signed-off-by: Marton Balint <cus@passwd.hu> --- Changelog | 1 + doc/bitstream_filters.texi | 30 ++++++ libavcodec/Makefile | 1 + libavcodec/bitstream_filters.c | 1 + libavcodec/pcm_rechunk_bsf.c | 204 +++++++++++++++++++++++++++++++++++++++++ libavcodec/version.h | 2 +- 6 files changed, 238 insertions(+), 1 deletion(-) create mode 100644 libavcodec/pcm_rechunk_bsf.c