Signed-off-by: Marton Balint <c...@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
diff --git a/Changelog b/Changelog index d9fcd8bb0a..6b0c911279 100644 --- a/Changelog +++ b/Changelog @@ -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: diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 8fe5b3ad75..70c276feed 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -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. diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 88944d9a3a..35968bdaf7 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -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 diff --git a/libavcodec/bitstream_filters.c b/libavcodec/bitstream_filters.c index 6b5ffe4d70..9e701191f8 100644 --- a/libavcodec/bitstream_filters.c +++ b/libavcodec/bitstream_filters.c @@ -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; diff --git a/libavcodec/pcm_rechunk_bsf.c b/libavcodec/pcm_rechunk_bsf.c new file mode 100644 index 0000000000..ca5c72c304 --- /dev/null +++ b/libavcodec/pcm_rechunk_bsf.c @@ -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, +}; diff --git a/libavcodec/version.h b/libavcodec/version.h index 8cff2e855b..ad85fb15e5 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -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, \ -- 2.16.4 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".