libavfilter/af_asyncts.c - third_party/ffmpeg - Git at Google

 /*
  * This file is part of Libav.
  *
  * Libav 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.
  *
  * Libav 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 Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 #include "libavresample/avresample.h"
 #include "libavutil/audio_fifo.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"

 #include "audio.h"
 #include "avfilter.h"

 typedef struct ASyncContext {
     const AVClass *class;

     AVAudioResampleContext *avr;
     int64_t pts;            ///< timestamp in samples of the first sample in fifo
     int min_delta;          ///< pad/trim min threshold in samples

     /* options */
     int resample;
     float min_delta_sec;
     int max_comp;
 } ASyncContext;

 #define OFFSET(x) offsetof(ASyncContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption options[] = {
     { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample),      AV_OPT_TYPE_INT,   { 0 },   0, 1,       A },
     { "min_delta",  "Minimum difference between timestamps and audio data "
                     "(in seconds) to trigger padding/trimmin the data.",        OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { 0.1 }, 0, INT_MAX, A },
     { "max_comp",   "Maximum compensation in samples per second.",              OFFSET(max_comp),      AV_OPT_TYPE_INT,   { 500 }, 0, INT_MAX, A },
     { NULL },
 };

 static const AVClass async_class = {
     .class_name = "asyncts filter",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };

 static int init(AVFilterContext *ctx, const char *args, void *opaque)
 {
     ASyncContext *s = ctx->priv;
     int ret;

     s->class = &async_class;
     av_opt_set_defaults(s);

     if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
         av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
         return ret;
     }
     av_opt_free(s);

     s->pts = AV_NOPTS_VALUE;

     return 0;
 }

 static void uninit(AVFilterContext *ctx)
 {
     ASyncContext *s = ctx->priv;

     if (s->avr) {
         avresample_close(s->avr);
         avresample_free(&s->avr);
     }
 }

 static int config_props(AVFilterLink *link)
 {
     ASyncContext *s = link->src->priv;
     int ret;

     s->min_delta = s->min_delta_sec * link->sample_rate;
     link->time_base = (AVRational){1, link->sample_rate};

     s->avr = avresample_alloc_context();
     if (!s->avr)
         return AVERROR(ENOMEM);

     av_opt_set_int(s->avr,  "in_channel_layout", link->channel_layout, 0);
     av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);
     av_opt_set_int(s->avr,  "in_sample_fmt",     link->format,         0);
     av_opt_set_int(s->avr, "out_sample_fmt",     link->format,         0);
     av_opt_set_int(s->avr,  "in_sample_rate",    link->sample_rate,    0);
     av_opt_set_int(s->avr, "out_sample_rate",    link->sample_rate,    0);

     if (s->resample)
         av_opt_set_int(s->avr, "force_resampling", 1, 0);

     if ((ret = avresample_open(s->avr)) < 0)
         return ret;

     return 0;
 }

 static int request_frame(AVFilterLink *link)
 {
     AVFilterContext *ctx = link->src;
     ASyncContext      *s = ctx->priv;
     int ret = avfilter_request_frame(ctx->inputs[0]);
     int nb_samples;

     /* flush the fifo */
     if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
         AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
                                                      nb_samples);
         if (!buf)
             return AVERROR(ENOMEM);
         avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
                            nb_samples, NULL, 0, 0);
         buf->pts = s->pts;
         ff_filter_samples(link, buf);
         return 0;
     }

     return ret;
 }

 static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
 {
     avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
                        buf->linesize[0], buf->audio->nb_samples);
     avfilter_unref_buffer(buf);
 }

 /* get amount of data currently buffered, in samples */
 static int64_t get_delay(ASyncContext *s)
 {
     return avresample_available(s->avr) + avresample_get_delay(s->avr);
 }

 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     ASyncContext       *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
     int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
                   av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
     int out_size;
     int64_t delta;

     /* buffer data until we get the first timestamp */
     if (s->pts == AV_NOPTS_VALUE) {
         if (pts != AV_NOPTS_VALUE) {
             s->pts = pts - get_delay(s);
         }
         write_to_fifo(s, buf);
         return;
     }

     /* now wait for the next timestamp */
     if (pts == AV_NOPTS_VALUE) {
         write_to_fifo(s, buf);
         return;
     }

     /* when we have two timestamps, compute how many samples would we have
      * to add/remove to get proper sync between data and timestamps */
     delta    = pts - s->pts - get_delay(s);
     out_size = avresample_available(s->avr);

     if (labs(delta) > s->min_delta) {
         av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
         out_size += delta;
     } else {
         if (s->resample) {
             int comp = av_clip(delta, -s->max_comp, s->max_comp);
             av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
             avresample_set_compensation(s->avr, delta, inlink->sample_rate);
         }
         delta = 0;
     }

     if (out_size > 0) {
         AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
                                                          out_size);
         if (!buf_out)
             return;

         avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
         buf_out->pts = s->pts;

         if (delta > 0) {
             av_samples_set_silence(buf_out->extended_data, out_size - delta,
                                    delta, nb_channels, buf->format);
         }
         ff_filter_samples(outlink, buf_out);
     } else {
         av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
                "whole buffer.\n");
     }

     /* drain any remaining buffered data */
     avresample_read(s->avr, NULL, avresample_available(s->avr));

     s->pts = pts - avresample_get_delay(s->avr);
     avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
                        buf->linesize[0], buf->audio->nb_samples);
     avfilter_unref_buffer(buf);
 }

 AVFilter avfilter_af_asyncts = {
     .name        = "asyncts",
     .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),

     .init        = init,
     .uninit      = uninit,

     .priv_size   = sizeof(ASyncContext),

     .inputs      = (const AVFilterPad[]) {{ .name           = "default",
                                             .type           = AVMEDIA_TYPE_AUDIO,
                                             .filter_samples = filter_samples },
                                           { NULL }},
     .outputs     = (const AVFilterPad[]) {{ .name           = "default",
                                             .type           = AVMEDIA_TYPE_AUDIO,
                                             .config_props   = config_props,
                                             .request_frame  = request_frame },
                                           { NULL }},
 };
	/*
	* This file is part of Libav.
	*
	* Libav 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.
	*
	* Libav 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 Libav; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	#include "libavresample/avresample.h"
	#include "libavutil/audio_fifo.h"
	#include "libavutil/mathematics.h"
	#include "libavutil/opt.h"
	#include "libavutil/samplefmt.h"

	#include "audio.h"
	#include "avfilter.h"

	typedef struct ASyncContext {
	const AVClass *class;

	AVAudioResampleContext *avr;
	int64_t pts; ///< timestamp in samples of the first sample in fifo
	int min_delta; ///< pad/trim min threshold in samples

	/* options */
	int resample;
	float min_delta_sec;
	int max_comp;
	} ASyncContext;

	#define OFFSET(x) offsetof(ASyncContext, x)
	#define A AV_OPT_FLAG_AUDIO_PARAM
	static const AVOption options[] = {
	{ "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { 0 }, 0, 1, A },
	{ "min_delta", "Minimum difference between timestamps and audio data "
	"(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { 0.1 }, 0, INT_MAX, A },
	{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { 500 }, 0, INT_MAX, A },
	{ NULL },
	};

	static const AVClass async_class = {
	.class_name = "asyncts filter",
	.item_name = av_default_item_name,
	.option = options,
	.version = LIBAVUTIL_VERSION_INT,
	};

	static int init(AVFilterContext ctx, const char args, void *opaque)
	{
	ASyncContext *s = ctx->priv;
	int ret;

	s->class = &async_class;
	av_opt_set_defaults(s);

	if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
	av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
	return ret;
	}
	av_opt_free(s);

	s->pts = AV_NOPTS_VALUE;

	return 0;
	}

	static void uninit(AVFilterContext *ctx)
	{
	ASyncContext *s = ctx->priv;

	if (s->avr) {
	avresample_close(s->avr);
	avresample_free(&s->avr);
	}
	}

	static int config_props(AVFilterLink *link)
	{
	ASyncContext *s = link->src->priv;
	int ret;

	s->min_delta = s->min_delta_sec * link->sample_rate;
	link->time_base = (AVRational){1, link->sample_rate};

	s->avr = avresample_alloc_context();
	if (!s->avr)
	return AVERROR(ENOMEM);

	av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0);
	av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);
	av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0);
	av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0);
	av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0);
	av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0);

	if (s->resample)
	av_opt_set_int(s->avr, "force_resampling", 1, 0);

	if ((ret = avresample_open(s->avr)) < 0)
	return ret;

	return 0;
	}

	static int request_frame(AVFilterLink *link)
	{
	AVFilterContext *ctx = link->src;
	ASyncContext *s = ctx->priv;
	int ret = avfilter_request_frame(ctx->inputs[0]);
	int nb_samples;

	/* flush the fifo */
	if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
	AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
	nb_samples);
	if (!buf)
	return AVERROR(ENOMEM);
	avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
	nb_samples, NULL, 0, 0);
	buf->pts = s->pts;
	ff_filter_samples(link, buf);
	return 0;
	}

	return ret;
	}

	static void write_to_fifo(ASyncContext s, AVFilterBufferRef buf)
	{
	avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
	buf->linesize[0], buf->audio->nb_samples);
	avfilter_unref_buffer(buf);
	}

	/* get amount of data currently buffered, in samples */
	static int64_t get_delay(ASyncContext *s)
	{
	return avresample_available(s->avr) + avresample_get_delay(s->avr);
	}

	static void filter_samples(AVFilterLink inlink, AVFilterBufferRef buf)
	{
	AVFilterContext *ctx = inlink->dst;
	ASyncContext *s = ctx->priv;
	AVFilterLink *outlink = ctx->outputs[0];
	int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
	int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
	av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
	int out_size;
	int64_t delta;

	/* buffer data until we get the first timestamp */
	if (s->pts == AV_NOPTS_VALUE) {
	if (pts != AV_NOPTS_VALUE) {
	s->pts = pts - get_delay(s);
	}
	write_to_fifo(s, buf);
	return;
	}

	/* now wait for the next timestamp */
	if (pts == AV_NOPTS_VALUE) {
	write_to_fifo(s, buf);
	return;
	}

	/* when we have two timestamps, compute how many samples would we have
	* to add/remove to get proper sync between data and timestamps */
	delta = pts - s->pts - get_delay(s);
	out_size = avresample_available(s->avr);

	if (labs(delta) > s->min_delta) {
	av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
	out_size += delta;
	} else {
	if (s->resample) {
	int comp = av_clip(delta, -s->max_comp, s->max_comp);
	av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
	avresample_set_compensation(s->avr, delta, inlink->sample_rate);
	}
	delta = 0;
	}

	if (out_size > 0) {
	AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
	out_size);
	if (!buf_out)
	return;

	avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
	buf_out->pts = s->pts;

	if (delta > 0) {
	av_samples_set_silence(buf_out->extended_data, out_size - delta,
	delta, nb_channels, buf->format);
	}
	ff_filter_samples(outlink, buf_out);
	} else {
	av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
	"whole buffer.\n");
	}

	/* drain any remaining buffered data */
	avresample_read(s->avr, NULL, avresample_available(s->avr));

	s->pts = pts - avresample_get_delay(s->avr);
	avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
	buf->linesize[0], buf->audio->nb_samples);
	avfilter_unref_buffer(buf);
	}

	AVFilter avfilter_af_asyncts = {
	.name = "asyncts",
	.description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),

	.init = init,
	.uninit = uninit,

	.priv_size = sizeof(ASyncContext),

	.inputs = (const AVFilterPad[]) {{ .name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.filter_samples = filter_samples },
	{ NULL }},
	.outputs = (const AVFilterPad[]) {{ .name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.config_props = config_props,
	.request_frame = request_frame },
	{ NULL }},
	};