doc/examples/filtering_audio.c - manifest_repos/ffmpeg - Git at Google

 /*
  * Copyright (c) 2010 Nicolas George
  * Copyright (c) 2011 Stefano Sabatini
  * Copyright (c) 2012 Clément Bœsch
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy
  * of this software and associated documentation files (the "Software"), to deal
  * in the Software without restriction, including without limitation the rights
  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  * copies of the Software, and to permit persons to whom the Software is
  * furnished to do so, subject to the following conditions:
  *
  * The above copyright notice and this permission notice shall be included in
  * all copies or substantial portions of the Software.
  *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */

 /**
  * @file
  * API example for audio decoding and filtering
  * @example filtering_audio.c
  */

 #include <unistd.h>

 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>

 static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
 static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
 AVFilterContext *buffersink_ctx;
 AVFilterContext *buffersrc_ctx;
 AVFilterGraph *filter_graph;
 static int audio_stream_index = -1;

 static int open_input_file(const char *filename)
 {
     int ret;
     AVCodec *dec;

     if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
         return ret;
     }

     if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
         return ret;
     }

     /* select the audio stream */
     ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
         return ret;
     }
     audio_stream_index = ret;

     /* create decoding context */
     dec_ctx = avcodec_alloc_context3(dec);
     if (!dec_ctx)
         return AVERROR(ENOMEM);
     avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);

     /* init the audio decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
         return ret;
     }

     return 0;
 }

 static int init_filters(const char *filters_descr)
 {
     char args[512];
     int ret = 0;
     const AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
     const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
     static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
     static const int out_sample_rates[] = { 8000, -1 };
     const AVFilterLink *outlink;
     AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

     filter_graph = avfilter_graph_alloc();
     if (!outputs || !inputs || !filter_graph) {
         ret = AVERROR(ENOMEM);
         goto end;
     }

     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
     if (!dec_ctx->channel_layout)
         dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
     snprintf(args, sizeof(args),
             "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
              time_base.num, time_base.den, dec_ctx->sample_rate,
              av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
     ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
         goto end;
     }

     /* buffer audio sink: to terminate the filter chain. */
     ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
                                        NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
         goto end;
     }

     ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
         goto end;
     }

     ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
         goto end;
     }

     ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
         goto end;
     }

     /*
      * Set the endpoints for the filter graph. The filter_graph will
      * be linked to the graph described by filters_descr.
      */

     /*
      * The buffer source output must be connected to the input pad of
      * the first filter described by filters_descr; since the first
      * filter input label is not specified, it is set to "in" by
      * default.
      */
     outputs->name       = av_strdup("in");
     outputs->filter_ctx = buffersrc_ctx;
     outputs->pad_idx    = 0;
     outputs->next       = NULL;

     /*
      * The buffer sink input must be connected to the output pad of
      * the last filter described by filters_descr; since the last
      * filter output label is not specified, it is set to "out" by
      * default.
      */
     inputs->name       = av_strdup("out");
     inputs->filter_ctx = buffersink_ctx;
     inputs->pad_idx    = 0;
     inputs->next       = NULL;

     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                         &inputs, &outputs, NULL)) < 0)
         goto end;

     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
         goto end;

     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
     outlink = buffersink_ctx->inputs[0];
     av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
     av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
            (int)outlink->sample_rate,
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
            args);

 end:
     avfilter_inout_free(&inputs);
     avfilter_inout_free(&outputs);

     return ret;
 }

 static void print_frame(const AVFrame *frame)
 {
     const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
     const uint16_t *p     = (uint16_t*)frame->data[0];
     const uint16_t *p_end = p + n;

     while (p < p_end) {
         fputc(*p    & 0xff, stdout);
         fputc(*p>>8 & 0xff, stdout);
         p++;
     }
     fflush(stdout);
 }

 int main(int argc, char **argv)
 {
     int ret;
     AVPacket packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();

     if (!frame || !filt_frame) {
         perror("Could not allocate frame");
         exit(1);
     }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
         exit(1);
     }

     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
     if ((ret = init_filters(filter_descr)) < 0)
         goto end;

     /* read all packets */
     while (1) {
         if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
             break;

         if (packet.stream_index == audio_stream_index) {
             ret = avcodec_send_packet(dec_ctx, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                 break;
             }

             while (ret >= 0) {
                 ret = avcodec_receive_frame(dec_ctx, frame);
                 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                     break;
                 } else if (ret < 0) {
                     av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
                     goto end;
                 }

                 if (ret >= 0) {
                     /* push the audio data from decoded frame into the filtergraph */
                     if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
                         av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                         break;
                     }

                     /* pull filtered audio from the filtergraph */
                     while (1) {
                         ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                         if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                             break;
                         if (ret < 0)
                             goto end;
                         print_frame(filt_frame);
                         av_frame_unref(filt_frame);
                     }
                     av_frame_unref(frame);
                 }
             }
         }
         av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
     avcodec_free_context(&dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);

     if (ret < 0 && ret != AVERROR_EOF) {
         fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
         exit(1);
     }

     exit(0);
 }
	/*
	* Copyright (c) 2010 Nicolas George
	* Copyright (c) 2011 Stefano Sabatini
	* Copyright (c) 2012 Clément Bœsch
	*
	* Permission is hereby granted, free of charge, to any person obtaining a copy
	* of this software and associated documentation files (the "Software"), to deal
	* in the Software without restriction, including without limitation the rights
	* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
	* copies of the Software, and to permit persons to whom the Software is
	* furnished to do so, subject to the following conditions:
	*
	* The above copyright notice and this permission notice shall be included in
	* all copies or substantial portions of the Software.
	*
	* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
	* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
	* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
	* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
	* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
	* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
	* THE SOFTWARE.
	*/

	/**
	* @file
	* API example for audio decoding and filtering
	* @example filtering_audio.c
	*/

	#include <unistd.h>

	#include <libavcodec/avcodec.h>
	#include <libavformat/avformat.h>
	#include <libavfilter/buffersink.h>
	#include <libavfilter/buffersrc.h>
	#include <libavutil/opt.h>

	static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
	static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";

	static AVFormatContext *fmt_ctx;
	static AVCodecContext *dec_ctx;
	AVFilterContext *buffersink_ctx;
	AVFilterContext *buffersrc_ctx;
	AVFilterGraph *filter_graph;
	static int audio_stream_index = -1;

	static int open_input_file(const char *filename)
	{
	int ret;
	AVCodec *dec;

	if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
	return ret;
	}

	if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
	return ret;
	}

	/* select the audio stream */
	ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
	return ret;
	}
	audio_stream_index = ret;

	/* create decoding context */
	dec_ctx = avcodec_alloc_context3(dec);
	if (!dec_ctx)
	return AVERROR(ENOMEM);
	avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);

	/* init the audio decoder */
	if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
	return ret;
	}

	return 0;
	}

	static int init_filters(const char *filters_descr)
	{
	char args[512];
	int ret = 0;
	const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
	const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
	AVFilterInOut *outputs = avfilter_inout_alloc();
	AVFilterInOut *inputs = avfilter_inout_alloc();
	static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
	static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
	static const int out_sample_rates[] = { 8000, -1 };
	const AVFilterLink *outlink;
	AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

	filter_graph = avfilter_graph_alloc();
	if (!outputs \|\| !inputs \|\| !filter_graph) {
	ret = AVERROR(ENOMEM);
	goto end;
	}

	/* buffer audio source: the decoded frames from the decoder will be inserted here. */
	if (!dec_ctx->channel_layout)
	dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
	snprintf(args, sizeof(args),
	"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
	time_base.num, time_base.den, dec_ctx->sample_rate,
	av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
	ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
	args, NULL, filter_graph);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
	goto end;
	}

	/* buffer audio sink: to terminate the filter chain. */
	ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
	NULL, NULL, filter_graph);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
	goto end;
	}

	ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
	AV_OPT_SEARCH_CHILDREN);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
	goto end;
	}

	ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
	AV_OPT_SEARCH_CHILDREN);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
	goto end;
	}

	ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
	AV_OPT_SEARCH_CHILDREN);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
	goto end;
	}

	/*
	* Set the endpoints for the filter graph. The filter_graph will
	* be linked to the graph described by filters_descr.
	*/

	/*
	* The buffer source output must be connected to the input pad of
	* the first filter described by filters_descr; since the first
	* filter input label is not specified, it is set to "in" by
	* default.
	*/
	outputs->name = av_strdup("in");
	outputs->filter_ctx = buffersrc_ctx;
	outputs->pad_idx = 0;
	outputs->next = NULL;

	/*
	* The buffer sink input must be connected to the output pad of
	* the last filter described by filters_descr; since the last
	* filter output label is not specified, it is set to "out" by
	* default.
	*/
	inputs->name = av_strdup("out");
	inputs->filter_ctx = buffersink_ctx;
	inputs->pad_idx = 0;
	inputs->next = NULL;

	if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
	&inputs, &outputs, NULL)) < 0)
	goto end;

	if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
	goto end;

	/* Print summary of the sink buffer
	* Note: args buffer is reused to store channel layout string */
	outlink = buffersink_ctx->inputs[0];
	av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
	av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
	(int)outlink->sample_rate,
	(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
	args);

	end:
	avfilter_inout_free(&inputs);
	avfilter_inout_free(&outputs);

	return ret;
	}

	static void print_frame(const AVFrame *frame)
	{
	const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
	const uint16_t p = (uint16_t)frame->data[0];
	const uint16_t *p_end = p + n;

	while (p < p_end) {
	fputc(*p & 0xff, stdout);
	fputc(*p>>8 & 0xff, stdout);
	p++;
	}
	fflush(stdout);
	}

	int main(int argc, char **argv)
	{
	int ret;
	AVPacket packet;
	AVFrame *frame = av_frame_alloc();
	AVFrame *filt_frame = av_frame_alloc();

	if (!frame \|\| !filt_frame) {
	perror("Could not allocate frame");
	exit(1);
	}
	if (argc != 2) {
	fprintf(stderr, "Usage: %s file \| %s\n", argv[0], player);
	exit(1);
	}

	if ((ret = open_input_file(argv[1])) < 0)
	goto end;
	if ((ret = init_filters(filter_descr)) < 0)
	goto end;

	/* read all packets */
	while (1) {
	if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
	break;

	if (packet.stream_index == audio_stream_index) {
	ret = avcodec_send_packet(dec_ctx, &packet);
	if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
	break;
	}

	while (ret >= 0) {
	ret = avcodec_receive_frame(dec_ctx, frame);
	if (ret == AVERROR(EAGAIN) \|\| ret == AVERROR_EOF) {
	break;
	} else if (ret < 0) {
	av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
	goto end;
	}

	if (ret >= 0) {
	/* push the audio data from decoded frame into the filtergraph */
	if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
	av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
	break;
	}

	/* pull filtered audio from the filtergraph */
	while (1) {
	ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
	if (ret == AVERROR(EAGAIN) \|\| ret == AVERROR_EOF)
	break;
	if (ret < 0)
	goto end;
	print_frame(filt_frame);
	av_frame_unref(filt_frame);
	}
	av_frame_unref(frame);
	}
	}
	}
	av_packet_unref(&packet);
	}
	end:
	avfilter_graph_free(&filter_graph);
	avcodec_free_context(&dec_ctx);
	avformat_close_input(&fmt_ctx);
	av_frame_free(&frame);
	av_frame_free(&filt_frame);

	if (ret < 0 && ret != AVERROR_EOF) {
	fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
	exit(1);
	}

	exit(0);
	}