libavcodec/pcm_rechunk_bsf.c - manifest_repos/ffmpeg - Git at Google

 /*
  * 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_internal.h"
 #include "libavutil/avassert.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;
 } 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();
     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;
 }

 static int send_packet(PCMContext *s, int nb_samples, AVPacket *pkt)
 {
     pkt->duration = nb_samples;
     s->n++;
     return 0;
 }

 static void drain_packet(AVPacket *pkt, int drain_data, int drain_samples)
 {
     pkt->size -= drain_data;
     pkt->data += drain_data;
     if (pkt->dts != AV_NOPTS_VALUE)
         pkt->dts += drain_samples;
     if (pkt->pts != AV_NOPTS_VALUE)
         pkt->pts += drain_samples;
 }

 static int get_next_nb_samples(AVBSFContext *ctx)
 {
     PCMContext *s = ctx->priv_data;
     if (s->frame_rate.num) {
         AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
         return av_rescale_q(s->n + 1, sr, s->frame_rate) - av_rescale_q(s->n, sr, s->frame_rate);
     } else {
         return s->nb_out_samples;
     }
 }

 static int rechunk_filter(AVBSFContext *ctx, AVPacket *pkt)
 {
     PCMContext *s = ctx->priv_data;
     int nb_samples = get_next_nb_samples(ctx);
     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;
                 drain_packet(s->in_pkt, drain, drain / s->sample_size);
                 if (!s->in_pkt->size)
                     av_packet_unref(s->in_pkt);
                 if (s->out_pkt->size == data_size) {
                     av_packet_move_ref(pkt, s->out_pkt);
                     return send_packet(s, nb_samples, 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;
                 drain_packet(s->in_pkt, data_size, nb_samples);
                 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);
         }
         if (ret >= 0)
             av_packet_rescale_ts(s->in_pkt, ctx->time_base_in, ctx->time_base_out);
     } 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,
 };
	/*
	* 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_internal.h"
	#include "libavutil/avassert.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;
	} 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();
	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;
	}

	static int send_packet(PCMContext s, int nb_samples, AVPacket pkt)
	{
	pkt->duration = nb_samples;
	s->n++;
	return 0;
	}

	static void drain_packet(AVPacket *pkt, int drain_data, int drain_samples)
	{
	pkt->size -= drain_data;
	pkt->data += drain_data;
	if (pkt->dts != AV_NOPTS_VALUE)
	pkt->dts += drain_samples;
	if (pkt->pts != AV_NOPTS_VALUE)
	pkt->pts += drain_samples;
	}

	static int get_next_nb_samples(AVBSFContext *ctx)
	{
	PCMContext *s = ctx->priv_data;
	if (s->frame_rate.num) {
	AVRational sr = av_make_q(ctx->par_in->sample_rate, 1);
	return av_rescale_q(s->n + 1, sr, s->frame_rate) - av_rescale_q(s->n, sr, s->frame_rate);
	} else {
	return s->nb_out_samples;
	}
	}

	static int rechunk_filter(AVBSFContext ctx, AVPacket pkt)
	{
	PCMContext *s = ctx->priv_data;
	int nb_samples = get_next_nb_samples(ctx);
	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;
	drain_packet(s->in_pkt, drain, drain / s->sample_size);
	if (!s->in_pkt->size)
	av_packet_unref(s->in_pkt);
	if (s->out_pkt->size == data_size) {
	av_packet_move_ref(pkt, s->out_pkt);
	return send_packet(s, nb_samples, 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;
	drain_packet(s->in_pkt, data_size, nb_samples);
	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);
	}
	if (ret >= 0)
	av_packet_rescale_ts(s->in_pkt, ctx->time_base_in, ctx->time_base_out);
	} 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,
	};