mirror of https://git.ffmpeg.org/ffmpeg.git
295 lines
10 KiB
C
295 lines
10 KiB
C
/*
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* Copyright (c) 2010 Nicolas George
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* Copyright (c) 2011 Stefano Sabatini
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* Copyright (c) 2012 Clément Bœsch
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/**
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* @file
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* API example for audio decoding and filtering
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* @example filtering_audio.c
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*/
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#include <unistd.h>
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavfilter/buffersink.h>
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#include <libavfilter/buffersrc.h>
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#include <libavutil/channel_layout.h>
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#include <libavutil/opt.h>
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static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
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static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
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static AVFormatContext *fmt_ctx;
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static AVCodecContext *dec_ctx;
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AVFilterContext *buffersink_ctx;
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AVFilterContext *buffersrc_ctx;
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AVFilterGraph *filter_graph;
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static int audio_stream_index = -1;
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static int open_input_file(const char *filename)
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{
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const AVCodec *dec;
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int ret;
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if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
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return ret;
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}
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if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
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return ret;
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}
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/* select the audio stream */
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ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
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return ret;
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}
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audio_stream_index = ret;
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/* create decoding context */
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dec_ctx = avcodec_alloc_context3(dec);
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if (!dec_ctx)
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return AVERROR(ENOMEM);
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avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
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/* init the audio decoder */
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if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
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return ret;
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}
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return 0;
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}
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static int init_filters(const char *filters_descr)
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{
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char args[512];
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int ret = 0;
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const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
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const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
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AVFilterInOut *outputs = avfilter_inout_alloc();
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AVFilterInOut *inputs = avfilter_inout_alloc();
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static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
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static const int out_sample_rates[] = { 8000, -1 };
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const AVFilterLink *outlink;
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AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
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filter_graph = avfilter_graph_alloc();
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if (!outputs || !inputs || !filter_graph) {
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ret = AVERROR(ENOMEM);
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goto end;
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}
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/* buffer audio source: the decoded frames from the decoder will be inserted here. */
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if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
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av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
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ret = snprintf(args, sizeof(args),
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"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
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time_base.num, time_base.den, dec_ctx->sample_rate,
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av_get_sample_fmt_name(dec_ctx->sample_fmt));
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av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
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ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
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args, NULL, filter_graph);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
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goto end;
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}
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/* buffer audio sink: to terminate the filter chain. */
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ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
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NULL, NULL, filter_graph);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
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goto end;
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}
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ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
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AV_OPT_SEARCH_CHILDREN);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
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goto end;
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}
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ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
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AV_OPT_SEARCH_CHILDREN);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
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goto end;
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}
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ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
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AV_OPT_SEARCH_CHILDREN);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
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goto end;
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}
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/*
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* Set the endpoints for the filter graph. The filter_graph will
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* be linked to the graph described by filters_descr.
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*/
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/*
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* The buffer source output must be connected to the input pad of
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* the first filter described by filters_descr; since the first
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* filter input label is not specified, it is set to "in" by
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* default.
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*/
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outputs->name = av_strdup("in");
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outputs->filter_ctx = buffersrc_ctx;
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outputs->pad_idx = 0;
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outputs->next = NULL;
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/*
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* The buffer sink input must be connected to the output pad of
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* the last filter described by filters_descr; since the last
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* filter output label is not specified, it is set to "out" by
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* default.
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*/
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inputs->name = av_strdup("out");
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inputs->filter_ctx = buffersink_ctx;
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inputs->pad_idx = 0;
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inputs->next = NULL;
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if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
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&inputs, &outputs, NULL)) < 0)
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goto end;
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if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
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goto end;
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/* Print summary of the sink buffer
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* Note: args buffer is reused to store channel layout string */
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outlink = buffersink_ctx->inputs[0];
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av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
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av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
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(int)outlink->sample_rate,
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(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
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args);
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end:
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avfilter_inout_free(&inputs);
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avfilter_inout_free(&outputs);
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return ret;
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}
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static void print_frame(const AVFrame *frame)
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{
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const int n = frame->nb_samples * frame->ch_layout.nb_channels;
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const uint16_t *p = (uint16_t*)frame->data[0];
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const uint16_t *p_end = p + n;
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while (p < p_end) {
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fputc(*p & 0xff, stdout);
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fputc(*p>>8 & 0xff, stdout);
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p++;
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}
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fflush(stdout);
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}
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int main(int argc, char **argv)
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{
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int ret;
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AVPacket *packet = av_packet_alloc();
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AVFrame *frame = av_frame_alloc();
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AVFrame *filt_frame = av_frame_alloc();
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if (!packet || !frame || !filt_frame) {
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fprintf(stderr, "Could not allocate frame or packet\n");
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exit(1);
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}
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if (argc != 2) {
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fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
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exit(1);
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}
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if ((ret = open_input_file(argv[1])) < 0)
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goto end;
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if ((ret = init_filters(filter_descr)) < 0)
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goto end;
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/* read all packets */
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while (1) {
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if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
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break;
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if (packet->stream_index == audio_stream_index) {
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ret = avcodec_send_packet(dec_ctx, packet);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
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break;
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}
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while (ret >= 0) {
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ret = avcodec_receive_frame(dec_ctx, frame);
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if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
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break;
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} else if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
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goto end;
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}
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if (ret >= 0) {
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/* push the audio data from decoded frame into the filtergraph */
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if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
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av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
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break;
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}
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/* pull filtered audio from the filtergraph */
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while (1) {
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ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
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if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
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break;
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if (ret < 0)
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goto end;
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print_frame(filt_frame);
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av_frame_unref(filt_frame);
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}
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av_frame_unref(frame);
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}
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}
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}
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av_packet_unref(packet);
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}
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end:
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avfilter_graph_free(&filter_graph);
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avcodec_free_context(&dec_ctx);
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avformat_close_input(&fmt_ctx);
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av_packet_free(&packet);
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av_frame_free(&frame);
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av_frame_free(&filt_frame);
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if (ret < 0 && ret != AVERROR_EOF) {
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fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
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exit(1);
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}
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exit(0);
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}
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