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+/*
+ * 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
+ */
+
+#include <unistd.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/avcodec.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+
+const char *filter_descr = "aresample=8000,aconvert=s16:mono";
+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 a audio stream in the input file\n");
+ return ret;
+ }
+ audio_stream_index = ret;
+ dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
+
+ /* 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;
+ AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
+ AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
+ AVFilterInOut *outputs = avfilter_inout_alloc();
+ AVFilterInOut *inputs = avfilter_inout_alloc();
+ const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+ AVABufferSinkParams *abuffersink_params;
+ const AVFilterLink *outlink;
+ AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
+
+ filter_graph = avfilter_graph_alloc();
+
+ /* 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");
+ return ret;
+ }
+
+ /* buffer audio sink: to terminate the filter chain. */
+ abuffersink_params = av_abuffersink_params_alloc();
+ abuffersink_params->sample_fmts = sample_fmts;
+ ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
+ NULL, abuffersink_params, filter_graph);
+ av_free(abuffersink_params);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
+ return ret;
+ }
+
+ /* Endpoints for the filter graph. */
+ outputs->name = av_strdup("in");
+ outputs->filter_ctx = buffersrc_ctx;
+ outputs->pad_idx = 0;
+ outputs->next = NULL;
+
+ inputs->name = av_strdup("out");
+ inputs->filter_ctx = buffersink_ctx;
+ inputs->pad_idx = 0;
+ inputs->next = NULL;
+
+ if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+ &inputs, &outputs, NULL)) < 0)
+ return ret;
+
+ if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+ return ret;
+
+ /* 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);
+
+ return 0;
+}
+
+static void print_samplesref(AVFilterBufferRef *samplesref)
+{
+ const AVFilterBufferRefAudioProps *props = samplesref->audio;
+ const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
+ const uint16_t *p = (uint16_t*)samplesref->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;
+ int got_frame;
+
+ if (argc != 2) {
+ fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
+ exit(1);
+ }
+
+ avcodec_register_all();
+ av_register_all();
+ avfilter_register_all();
+
+ if ((ret = open_input_file(argv[1])) < 0)
+ goto end;
+ if ((ret = init_filters(filter_descr)) < 0)
+ goto end;
+
+ /* read all packets */
+ while (1) {
+ AVFilterBufferRef *samplesref;
+ if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+ break;
+
+ if (packet.stream_index == audio_stream_index) {
+ avcodec_get_frame_defaults(&frame);
+ got_frame = 0;
+ ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
+ continue;
+ }
+
+ if (got_frame) {
+ /* push the audio data from decoded frame into the filtergraph */
+ if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 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_buffer_ref(buffersink_ctx, &samplesref, 0);
+ if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+ break;
+ if(ret < 0)
+ goto end;
+ if (samplesref) {
+ print_samplesref(samplesref);
+ avfilter_unref_bufferp(&samplesref);
+ }
+ }
+ }
+ }
+ av_free_packet(&packet);
+ }
+end:
+ avfilter_graph_free(&filter_graph);
+ if (dec_ctx)
+ avcodec_close(dec_ctx);
+ avformat_close_input(&fmt_ctx);
+
+ if (ret < 0 && ret != AVERROR_EOF) {
+ char buf[1024];
+ av_strerror(ret, buf, sizeof(buf));
+ fprintf(stderr, "Error occurred: %s\n", buf);
+ exit(1);
+ }
+
+ exit(0);
+}
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