doc/examples: add audio decoding/filtering example.

Mostly based on doc/examples/filtering.c. lavfi API is still limited to
"buffer feeding" instead of "frame feeding" at the moment, so this
example code sticks with it.

2 files changed, 246 insertions, 1 deletions

diff --git a/doc/examples/Makefile b/doc/examples/Makefile
index b4d299ff67..6b9902df6e 100644
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -3,7 +3,7 @@ FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
 CFLAGS+=-Wall $(shell pkg-config  --cflags $(FFMPEG_LIBS))
 LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
 
-EXAMPLES=decoding_encoding filtering metadata muxing
+EXAMPLES=decoding_encoding filtering filtering_audio metadata muxing
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c
new file mode 100644
index 0000000000..988dbfe3c2
--- /dev/null
+++ b/doc/examples/filtering_audio.c
@@ -0,0 +1,245 @@
+/*
+ * 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/asrc_abuffer.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/avcodec.h>
+#include <libavfilter/buffersink.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("abuffersink");
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    const int packing_fmts[]                = { AVFILTER_PACKED, -1 };
+    const int64_t *chlayouts                = avfilter_all_channel_layouts;
+    AVABufferSinkParams *abuffersink_params;
+    const AVFilterLink *outlink;
+
+    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), "%d:%d:0x%"PRIx64":packed",
+             dec_ctx->sample_rate, 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;
+    abuffersink_params->channel_layouts = chlayouts;
+    abuffersink_params->packing_fmts    = packing_fmts;
+    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
+                                       NULL, abuffersink_params, filter_graph);
+    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, filter_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);
+            av_free_packet(&packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
+                break;
+            }
+
+            if (got_frame) {
+                const int bps = av_get_bytes_per_sample(dec_ctx->sample_fmt);
+                const int decoded_data_size = frame.nb_samples * dec_ctx->channels * bps;
+
+                /* push the audio data from decoded frame into the filtergraph */
+                if (av_asrc_buffer_add_buffer(buffersrc_ctx,
+                                              frame.data[0],
+                                              decoded_data_size,
+                                              dec_ctx->sample_rate,
+                                              dec_ctx->sample_fmt,
+                                              dec_ctx->channel_layout,
+                                              0, frame.pts, 0) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
+                    break;
+                }
+
+                /* pull filtered audio from the filtergraph */
+                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
+                    av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
+                    if (samplesref) {
+                        print_samplesref(samplesref);
+                        avfilter_unref_buffer(samplesref);
+                    }
+                }
+            }
+        }
+    }
+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);
+}