[FFmpeg-devel] [PATCH v3 5/6] avcodec/pcm_rechunk_bsf: add bitstream filter to rechunk pcm audio
Marton Balint
cus at passwd.hu
Mon Apr 27 23:11:50 EEST 2020
On Sun, 26 Apr 2020, Andreas Rheinhardt wrote:
> Marton Balint:
>> Signed-off-by: Marton Balint <cus at passwd.hu>
>> ---
>> Changelog | 1 +
>> doc/bitstream_filters.texi | 30 ++++++
>> libavcodec/Makefile | 1 +
>> libavcodec/bitstream_filters.c | 1 +
>> libavcodec/pcm_rechunk_bsf.c | 206 +++++++++++++++++++++++++++++++++++++++++
>> libavcodec/version.h | 2 +-
>> 6 files changed, 240 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.
>>
>> + at 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.
>> +
>> + at table @option
>> + at 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.
>> +
>> + at 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.
>> +
>> + at 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
>
> "as close to the frame boundary as possible" or "as close as possible to
> the frame boundary"
>
>> +boundary as possible. Using this option has precedence over @option{nb_out_samples}.
>> + at 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.
>> + at example
>> +ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
>> + at 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..2a038fd79b
>> --- /dev/null
>> +++ b/libavcodec/pcm_rechunk_bsf.c
>> @@ -0,0 +1,206 @@
>> +/*
>> + * 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"
>
> I don't see where this header would be used -- your allocations are all
> performed implicitly by av_new_packet().
Ok, will remove.
>
>> +#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;
>> +
>> + if (ctx->par_in->channels <= 0 || ctx->par_in->sample_rate <= 0)
>> + return AVERROR(EINVAL);
>> +
>> + 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 - 1)
>> + return AVERROR(EINVAL);
>> +
>> + s->in_pkt = av_packet_alloc();
>> + s->out_pkt = av_packet_alloc();
>
> Could be aligned on "=".
Ok.
>
>> + 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;
>
> This implicitly presumes that the timebase is equal to the sample rate.
> Is this actually guaranteed? (Notice that you can set the output
> timebase as you want during init().)
Yes, and I set it in init() to the sample rate.
>
> And this filter does more than just repacketizing the samples: It also
> discards the timing of its input and makes up completely new timestamps
> and durations. This needs to be documented.
Ok, will do.
>
>> + 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);
>
> I would move this check in front of the check for whether out_pkt is
> full, so that there are not two places where in_pkt is unreferenced.
Yes, good point.
>
>> + 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, \
>>
> LGTM apart from the above comments.
Thanks, will send a new series anyway based on your comments.
Regards,
Marton
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