libavfilter/af_resample.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
  */

 /**
  * @file
  * sample format and channel layout conversion audio filter
  */

 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/common.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/opt.h"

 #include "libavresample/avresample.h"

 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
 #include "internal.h"

 typedef struct ResampleContext {
     AVAudioResampleContext *avr;

     int64_t next_pts;

     /* set by filter_frame() to signal an output frame to request_frame() */
     int got_output;
 } ResampleContext;

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

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

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterLink *inlink  = ctx->inputs[0];
     AVFilterLink *outlink = ctx->outputs[0];

     AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
     AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);
     AVFilterFormats        *in_samplerates  = ff_all_samplerates();
     AVFilterFormats        *out_samplerates = ff_all_samplerates();
     AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
     AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();

     ff_formats_ref(in_formats,  &inlink->out_formats);
     ff_formats_ref(out_formats, &outlink->in_formats);

     ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
     ff_formats_ref(out_samplerates, &outlink->in_samplerates);

     ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

     return 0;
 }

 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AVFilterLink *inlink = ctx->inputs[0];
     ResampleContext   *s = ctx->priv;
     char buf1[64], buf2[64];
     int ret;

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

     if (inlink->channel_layout == outlink->channel_layout &&
         inlink->sample_rate    == outlink->sample_rate    &&
         (inlink->format        == outlink->format ||
         (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&
          av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
          av_get_planar_sample_fmt(inlink->format) ==
          av_get_planar_sample_fmt(outlink->format))))
         return 0;

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

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

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

     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
     s->next_pts        = AV_NOPTS_VALUE;

     av_get_channel_layout_string(buf1, sizeof(buf1),
                                  -1, inlink ->channel_layout);
     av_get_channel_layout_string(buf2, sizeof(buf2),
                                  -1, outlink->channel_layout);
     av_log(ctx, AV_LOG_VERBOSE,
            "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);

     return 0;
 }

 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     ResampleContext   *s = ctx->priv;
     int ret = 0;

     s->got_output = 0;
     while (ret >= 0 && !s->got_output)
         ret = ff_request_frame(ctx->inputs[0]);

     /* flush the lavr delay buffer */
     if (ret == AVERROR_EOF && s->avr) {
         AVFilterBufferRef *buf;
         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
                                         outlink->sample_rate,
                                         ctx->inputs[0]->sample_rate,
                                         AV_ROUND_UP);

         if (!nb_samples)
             return ret;

         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
         if (!buf)
             return AVERROR(ENOMEM);

         ret = avresample_convert(s->avr, buf->extended_data,
                                  buf->linesize[0], nb_samples,
                                  NULL, 0, 0);
         if (ret <= 0) {
             avfilter_unref_buffer(buf);
             return (ret == 0) ? AVERROR_EOF : ret;
         }

         buf->pts = s->next_pts;
         return ff_filter_frame(outlink, buf);
     }
     return ret;
 }

 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     ResampleContext    *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     int ret;

     if (s->avr) {
         AVFilterBufferRef *buf_out;
         int delay, nb_samples;

         /* maximum possible samples lavr can output */
         delay      = avresample_get_delay(s->avr);
         nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
                                     outlink->sample_rate, inlink->sample_rate,
                                     AV_ROUND_UP);

         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
         if (!buf_out) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }

         ret     = avresample_convert(s->avr, buf_out->extended_data,
                                      buf_out->linesize[0], nb_samples,
                                      buf->extended_data, buf->linesize[0],
                                      buf->audio->nb_samples);
         if (ret <= 0) {
             avfilter_unref_buffer(buf_out);
             if (ret < 0)
                 goto fail;
         }

         av_assert0(!avresample_available(s->avr));

         if (s->next_pts == AV_NOPTS_VALUE) {
             if (buf->pts == AV_NOPTS_VALUE) {
                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
                        "assuming 0.\n");
                 s->next_pts = 0;
             } else
                 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
                                            outlink->time_base);
         }

         if (ret > 0) {
             buf_out->audio->nb_samples = ret;
             if (buf->pts != AV_NOPTS_VALUE) {
                 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
                                             outlink->time_base) -
                                av_rescale(delay, outlink->sample_rate,
                                           inlink->sample_rate);
             } else
                 buf_out->pts = s->next_pts;

             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;

             ret = ff_filter_frame(outlink, buf_out);
             s->got_output = 1;
         }

 fail:
         avfilter_unref_buffer(buf);
     } else {
         buf->format = outlink->format;
         ret = ff_filter_frame(outlink, buf);
         s->got_output = 1;
     }

     return ret;
 }

 static const AVFilterPad avfilter_af_resample_inputs[] = {
     {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,
         .filter_frame   = filter_frame,
         .min_perms      = AV_PERM_READ
     },
     { NULL }
 };

 static const AVFilterPad avfilter_af_resample_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = config_output,
         .request_frame = request_frame
     },
     { NULL }
 };

 AVFilter avfilter_af_resample = {
     .name          = "resample",
     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
     .priv_size     = sizeof(ResampleContext),

     .uninit         = uninit,
     .query_formats  = query_formats,

     .inputs    = avfilter_af_resample_inputs,
     .outputs   = avfilter_af_resample_outputs,
 };
	/*
	*
	* 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
	*/

	/**
	* @file
	* sample format and channel layout conversion audio filter
	*/

	#include "libavutil/avassert.h"
	#include "libavutil/avstring.h"
	#include "libavutil/common.h"
	#include "libavutil/mathematics.h"
	#include "libavutil/opt.h"

	#include "libavresample/avresample.h"

	#include "audio.h"
	#include "avfilter.h"
	#include "formats.h"
	#include "internal.h"

	typedef struct ResampleContext {
	AVAudioResampleContext *avr;

	int64_t next_pts;

	/* set by filter_frame() to signal an output frame to request_frame() */
	int got_output;
	} ResampleContext;

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

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

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterLink *inlink = ctx->inputs[0];
	AVFilterLink *outlink = ctx->outputs[0];

	AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
	AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
	AVFilterFormats *in_samplerates = ff_all_samplerates();
	AVFilterFormats *out_samplerates = ff_all_samplerates();
	AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
	AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();

	ff_formats_ref(in_formats, &inlink->out_formats);
	ff_formats_ref(out_formats, &outlink->in_formats);

	ff_formats_ref(in_samplerates, &inlink->out_samplerates);
	ff_formats_ref(out_samplerates, &outlink->in_samplerates);

	ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
	ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

	return 0;
	}

	static int config_output(AVFilterLink *outlink)
	{
	AVFilterContext *ctx = outlink->src;
	AVFilterLink *inlink = ctx->inputs[0];
	ResampleContext *s = ctx->priv;
	char buf1[64], buf2[64];
	int ret;

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

	if (inlink->channel_layout == outlink->channel_layout &&
	inlink->sample_rate == outlink->sample_rate &&
	(inlink->format == outlink->format \|\|
	(av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
	av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
	av_get_planar_sample_fmt(inlink->format) ==
	av_get_planar_sample_fmt(outlink->format))))
	return 0;

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

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

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

	outlink->time_base = (AVRational){ 1, outlink->sample_rate };
	s->next_pts = AV_NOPTS_VALUE;

	av_get_channel_layout_string(buf1, sizeof(buf1),
	-1, inlink ->channel_layout);
	av_get_channel_layout_string(buf2, sizeof(buf2),
	-1, outlink->channel_layout);
	av_log(ctx, AV_LOG_VERBOSE,
	"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
	av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
	av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);

	return 0;
	}

	static int request_frame(AVFilterLink *outlink)
	{
	AVFilterContext *ctx = outlink->src;
	ResampleContext *s = ctx->priv;
	int ret = 0;

	s->got_output = 0;
	while (ret >= 0 && !s->got_output)
	ret = ff_request_frame(ctx->inputs[0]);

	/* flush the lavr delay buffer */
	if (ret == AVERROR_EOF && s->avr) {
	AVFilterBufferRef *buf;
	int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
	outlink->sample_rate,
	ctx->inputs[0]->sample_rate,
	AV_ROUND_UP);

	if (!nb_samples)
	return ret;

	buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
	if (!buf)
	return AVERROR(ENOMEM);

	ret = avresample_convert(s->avr, buf->extended_data,
	buf->linesize[0], nb_samples,
	NULL, 0, 0);
	if (ret <= 0) {
	avfilter_unref_buffer(buf);
	return (ret == 0) ? AVERROR_EOF : ret;
	}

	buf->pts = s->next_pts;
	return ff_filter_frame(outlink, buf);
	}
	return ret;
	}

	static int filter_frame(AVFilterLink inlink, AVFilterBufferRef buf)
	{
	AVFilterContext *ctx = inlink->dst;
	ResampleContext *s = ctx->priv;
	AVFilterLink *outlink = ctx->outputs[0];
	int ret;

	if (s->avr) {
	AVFilterBufferRef *buf_out;
	int delay, nb_samples;

	/* maximum possible samples lavr can output */
	delay = avresample_get_delay(s->avr);
	nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
	outlink->sample_rate, inlink->sample_rate,
	AV_ROUND_UP);

	buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
	if (!buf_out) {
	ret = AVERROR(ENOMEM);
	goto fail;
	}

	ret = avresample_convert(s->avr, buf_out->extended_data,
	buf_out->linesize[0], nb_samples,
	buf->extended_data, buf->linesize[0],
	buf->audio->nb_samples);
	if (ret <= 0) {
	avfilter_unref_buffer(buf_out);
	if (ret < 0)
	goto fail;
	}

	av_assert0(!avresample_available(s->avr));

	if (s->next_pts == AV_NOPTS_VALUE) {
	if (buf->pts == AV_NOPTS_VALUE) {
	av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
	"assuming 0.\n");
	s->next_pts = 0;
	} else
	s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
	outlink->time_base);
	}

	if (ret > 0) {
	buf_out->audio->nb_samples = ret;
	if (buf->pts != AV_NOPTS_VALUE) {
	buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
	outlink->time_base) -
	av_rescale(delay, outlink->sample_rate,
	inlink->sample_rate);
	} else
	buf_out->pts = s->next_pts;

	s->next_pts = buf_out->pts + buf_out->audio->nb_samples;

	ret = ff_filter_frame(outlink, buf_out);
	s->got_output = 1;
	}

	fail:
	avfilter_unref_buffer(buf);
	} else {
	buf->format = outlink->format;
	ret = ff_filter_frame(outlink, buf);
	s->got_output = 1;
	}

	return ret;
	}

	static const AVFilterPad avfilter_af_resample_inputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.filter_frame = filter_frame,
	.min_perms = AV_PERM_READ
	},
	{ NULL }
	};

	static const AVFilterPad avfilter_af_resample_outputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.config_props = config_output,
	.request_frame = request_frame
	},
	{ NULL }
	};

	AVFilter avfilter_af_resample = {
	.name = "resample",
	.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
	.priv_size = sizeof(ResampleContext),

	.uninit = uninit,
	.query_formats = query_formats,

	.inputs = avfilter_af_resample_inputs,
	.outputs = avfilter_af_resample_outputs,
	};