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

 /*
  * Copyright (c) 2011 Stefano Sabatini
  * Copyright (c) 2011 Mina Nagy Zaki
  *
  * 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
  */

 /**
  * @file
  * resampling audio filter
  */

 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "libavutil/avassert.h"
 #include "libswresample/swresample.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"

 typedef struct {
     double ratio;
     struct SwrContext *swr;
     int64_t next_pts;
     int req_fullfilled;
 } AResampleContext;

 static av_cold int init(AVFilterContext *ctx, const char *args)
 {
     AResampleContext *aresample = ctx->priv;
     int ret = 0;
     char *argd = av_strdup(args);

     aresample->next_pts = AV_NOPTS_VALUE;
     aresample->swr = swr_alloc();
     if (!aresample->swr) {
         ret = AVERROR(ENOMEM);
         goto end;
     }

     if (args) {
         char *ptr = argd, *token;

         while (token = av_strtok(ptr, ":", &ptr)) {
             char *value;
             av_strtok(token, "=", &value);

             if (value) {
                 if ((ret = av_opt_set(aresample->swr, token, value, 0)) < 0)
                     goto end;
             } else {
                 int out_rate;
                 if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
                     goto end;
                 if ((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
                     goto end;
             }
         }
     }
 end:
     av_free(argd);
     return ret;
 }

 static av_cold void uninit(AVFilterContext *ctx)
 {
     AResampleContext *aresample = ctx->priv;
     swr_free(&aresample->swr);
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AResampleContext *aresample = ctx->priv;
     int out_rate                   = av_get_int(aresample->swr, "osr", NULL);
     uint64_t out_layout            = av_get_int(aresample->swr, "ocl", NULL);
     enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);

     AVFilterLink *inlink  = ctx->inputs[0];
     AVFilterLink *outlink = ctx->outputs[0];

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

     ff_formats_ref  (in_formats,      &inlink->out_formats);
     ff_formats_ref  (in_samplerates,  &inlink->out_samplerates);
     ff_channel_layouts_ref(in_layouts,      &inlink->out_channel_layouts);

     if(out_rate > 0) {
         out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
     } else {
         out_samplerates = ff_all_samplerates();
     }
     ff_formats_ref(out_samplerates, &outlink->in_samplerates);

     if(out_format != AV_SAMPLE_FMT_NONE) {
         out_formats = ff_make_format_list((int[]){ out_format, -1 });
     } else
         out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
     ff_formats_ref(out_formats, &outlink->in_formats);

     if(out_layout) {
         out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
     } else
         out_layouts = ff_all_channel_layouts();
     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

     return 0;
 }


 static int config_output(AVFilterLink *outlink)
 {
     int ret;
     AVFilterContext *ctx = outlink->src;
     AVFilterLink *inlink = ctx->inputs[0];
     AResampleContext *aresample = ctx->priv;
     int out_rate;
     uint64_t out_layout;
     enum AVSampleFormat out_format;
     char inchl_buf[128], outchl_buf[128];

     aresample->swr = swr_alloc_set_opts(aresample->swr,
                                         outlink->channel_layout, outlink->format, outlink->sample_rate,
                                         inlink->channel_layout, inlink->format, inlink->sample_rate,
                                         0, ctx);
     if (!aresample->swr)
         return AVERROR(ENOMEM);

     ret = swr_init(aresample->swr);
     if (ret < 0)
         return ret;

     out_rate   = av_get_int(aresample->swr, "osr", NULL);
     out_layout = av_get_int(aresample->swr, "ocl", NULL);
     out_format = av_get_int(aresample->swr, "osf", NULL);
     outlink->time_base = (AVRational) {1, out_rate};

     av_assert0(outlink->sample_rate == out_rate);
     av_assert0(outlink->channel_layout == out_layout);
     av_assert0(outlink->format == out_format);

     aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;

     av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  -1, inlink ->channel_layout);
     av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);

     av_log(ctx, AV_LOG_VERBOSE, "chl:%s fmt:%s r:%dHz -> chl:%s fmt:%s r:%dHz\n",
            inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
            outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
     return 0;
 }

 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
 {
     AResampleContext *aresample = inlink->dst->priv;
     const int n_in  = insamplesref->audio->nb_samples;
     int n_out       = n_in * aresample->ratio * 2 + 256;
     AVFilterLink *const outlink = inlink->dst->outputs[0];
     AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
     int ret;

     if(!outsamplesref)
         return AVERROR(ENOMEM);

     avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
     outsamplesref->format                = outlink->format;
     outsamplesref->audio->channels       = outlink->channels;
     outsamplesref->audio->channel_layout = outlink->channel_layout;
     outsamplesref->audio->sample_rate    = outlink->sample_rate;

     if(insamplesref->pts != AV_NOPTS_VALUE) {
         int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
         int64_t outpts= swr_next_pts(aresample->swr, inpts);
         aresample->next_pts =
         outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
     } else {
         outsamplesref->pts  = AV_NOPTS_VALUE;
     }
     n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
                                  (void *)insamplesref->extended_data, n_in);
     if (n_out <= 0) {
         avfilter_unref_buffer(outsamplesref);
         avfilter_unref_buffer(insamplesref);
         return 0;
     }

     outsamplesref->audio->nb_samples  = n_out;

     ret = ff_filter_frame(outlink, outsamplesref);
     aresample->req_fullfilled= 1;
     avfilter_unref_buffer(insamplesref);
     return ret;
 }

 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AResampleContext *aresample = ctx->priv;
     AVFilterLink *const inlink = outlink->src->inputs[0];
     int ret;

     aresample->req_fullfilled = 0;
     do{
         ret = ff_request_frame(ctx->inputs[0]);
     }while(!aresample->req_fullfilled && ret>=0);

     if (ret == AVERROR_EOF) {
         AVFilterBufferRef *outsamplesref;
         int n_out = 4096;

         outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
         if (!outsamplesref)
             return AVERROR(ENOMEM);
         n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
         if (n_out <= 0) {
             avfilter_unref_buffer(outsamplesref);
             return (n_out == 0) ? AVERROR_EOF : n_out;
         }

         outsamplesref->audio->sample_rate = outlink->sample_rate;
         outsamplesref->audio->nb_samples  = n_out;
 #if 0
         outsamplesref->pts = aresample->next_pts;
         if(aresample->next_pts != AV_NOPTS_VALUE)
             aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
 #else
         outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
         outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
 #endif

         ff_filter_frame(outlink, outsamplesref);
         return 0;
     }
     return ret;
 }

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

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

 AVFilter avfilter_af_aresample = {
     .name          = "aresample",
     .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
     .init          = init,
     .uninit        = uninit,
     .query_formats = query_formats,
     .priv_size     = sizeof(AResampleContext),
     .inputs        = aresample_inputs,
     .outputs       = aresample_outputs,
 };
	/*
	* Copyright (c) 2011 Stefano Sabatini
	* Copyright (c) 2011 Mina Nagy Zaki
	*
	* 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
	*/

	/**
	* @file
	* resampling audio filter
	*/

	#include "libavutil/avstring.h"
	#include "libavutil/channel_layout.h"
	#include "libavutil/opt.h"
	#include "libavutil/samplefmt.h"
	#include "libavutil/avassert.h"
	#include "libswresample/swresample.h"
	#include "avfilter.h"
	#include "audio.h"
	#include "internal.h"

	typedef struct {
	double ratio;
	struct SwrContext *swr;
	int64_t next_pts;
	int req_fullfilled;
	} AResampleContext;

	static av_cold int init(AVFilterContext ctx, const char args)
	{
	AResampleContext *aresample = ctx->priv;
	int ret = 0;
	char *argd = av_strdup(args);

	aresample->next_pts = AV_NOPTS_VALUE;
	aresample->swr = swr_alloc();
	if (!aresample->swr) {
	ret = AVERROR(ENOMEM);
	goto end;
	}

	if (args) {
	char ptr = argd, token;

	while (token = av_strtok(ptr, ":", &ptr)) {
	char *value;
	av_strtok(token, "=", &value);

	if (value) {
	if ((ret = av_opt_set(aresample->swr, token, value, 0)) < 0)
	goto end;
	} else {
	int out_rate;
	if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
	goto end;
	if ((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
	goto end;
	}
	}
	}
	end:
	av_free(argd);
	return ret;
	}

	static av_cold void uninit(AVFilterContext *ctx)
	{
	AResampleContext *aresample = ctx->priv;
	swr_free(&aresample->swr);
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AResampleContext *aresample = ctx->priv;
	int out_rate = av_get_int(aresample->swr, "osr", NULL);
	uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
	enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);

	AVFilterLink *inlink = ctx->inputs[0];
	AVFilterLink *outlink = ctx->outputs[0];

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

	ff_formats_ref (in_formats, &inlink->out_formats);
	ff_formats_ref (in_samplerates, &inlink->out_samplerates);
	ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);

	if(out_rate > 0) {
	out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
	} else {
	out_samplerates = ff_all_samplerates();
	}
	ff_formats_ref(out_samplerates, &outlink->in_samplerates);

	if(out_format != AV_SAMPLE_FMT_NONE) {
	out_formats = ff_make_format_list((int[]){ out_format, -1 });
	} else
	out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
	ff_formats_ref(out_formats, &outlink->in_formats);

	if(out_layout) {
	out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
	} else
	out_layouts = ff_all_channel_layouts();
	ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

	return 0;
	}


	static int config_output(AVFilterLink *outlink)
	{
	int ret;
	AVFilterContext *ctx = outlink->src;
	AVFilterLink *inlink = ctx->inputs[0];
	AResampleContext *aresample = ctx->priv;
	int out_rate;
	uint64_t out_layout;
	enum AVSampleFormat out_format;
	char inchl_buf[128], outchl_buf[128];

	aresample->swr = swr_alloc_set_opts(aresample->swr,
	outlink->channel_layout, outlink->format, outlink->sample_rate,
	inlink->channel_layout, inlink->format, inlink->sample_rate,
	0, ctx);
	if (!aresample->swr)
	return AVERROR(ENOMEM);

	ret = swr_init(aresample->swr);
	if (ret < 0)
	return ret;

	out_rate = av_get_int(aresample->swr, "osr", NULL);
	out_layout = av_get_int(aresample->swr, "ocl", NULL);
	out_format = av_get_int(aresample->swr, "osf", NULL);
	outlink->time_base = (AVRational) {1, out_rate};

	av_assert0(outlink->sample_rate == out_rate);
	av_assert0(outlink->channel_layout == out_layout);
	av_assert0(outlink->format == out_format);

	aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;

	av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), -1, inlink ->channel_layout);
	av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);

	av_log(ctx, AV_LOG_VERBOSE, "chl:%s fmt:%s r:%dHz -> chl:%s fmt:%s r:%dHz\n",
	inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
	outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
	return 0;
	}

	static int filter_frame(AVFilterLink inlink, AVFilterBufferRef insamplesref)
	{
	AResampleContext *aresample = inlink->dst->priv;
	const int n_in = insamplesref->audio->nb_samples;
	int n_out = n_in * aresample->ratio * 2 + 256;
	AVFilterLink *const outlink = inlink->dst->outputs[0];
	AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
	int ret;

	if(!outsamplesref)
	return AVERROR(ENOMEM);

	avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
	outsamplesref->format = outlink->format;
	outsamplesref->audio->channels = outlink->channels;
	outsamplesref->audio->channel_layout = outlink->channel_layout;
	outsamplesref->audio->sample_rate = outlink->sample_rate;

	if(insamplesref->pts != AV_NOPTS_VALUE) {
	int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
	int64_t outpts= swr_next_pts(aresample->swr, inpts);
	aresample->next_pts =
	outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
	} else {
	outsamplesref->pts = AV_NOPTS_VALUE;
	}
	n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
	(void *)insamplesref->extended_data, n_in);
	if (n_out <= 0) {
	avfilter_unref_buffer(outsamplesref);
	avfilter_unref_buffer(insamplesref);
	return 0;
	}

	outsamplesref->audio->nb_samples = n_out;

	ret = ff_filter_frame(outlink, outsamplesref);
	aresample->req_fullfilled= 1;
	avfilter_unref_buffer(insamplesref);
	return ret;
	}

	static int request_frame(AVFilterLink *outlink)
	{
	AVFilterContext *ctx = outlink->src;
	AResampleContext *aresample = ctx->priv;
	AVFilterLink *const inlink = outlink->src->inputs[0];
	int ret;

	aresample->req_fullfilled = 0;
	do{
	ret = ff_request_frame(ctx->inputs[0]);
	}while(!aresample->req_fullfilled && ret>=0);

	if (ret == AVERROR_EOF) {
	AVFilterBufferRef *outsamplesref;
	int n_out = 4096;

	outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
	if (!outsamplesref)
	return AVERROR(ENOMEM);
	n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
	if (n_out <= 0) {
	avfilter_unref_buffer(outsamplesref);
	return (n_out == 0) ? AVERROR_EOF : n_out;
	}

	outsamplesref->audio->sample_rate = outlink->sample_rate;
	outsamplesref->audio->nb_samples = n_out;
	#if 0
	outsamplesref->pts = aresample->next_pts;
	if(aresample->next_pts != AV_NOPTS_VALUE)
	aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
	#else
	outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
	outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
	#endif

	ff_filter_frame(outlink, outsamplesref);
	return 0;
	}
	return ret;
	}

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

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

	AVFilter avfilter_af_aresample = {
	.name = "aresample",
	.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
	.init = init,
	.uninit = uninit,
	.query_formats = query_formats,
	.priv_size = sizeof(AResampleContext),
	.inputs = aresample_inputs,
	.outputs = aresample_outputs,
	};