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.
This commit is contained in:
Clément Bœsch 2012-02-20 13:49:18 +01:00 committed by Clément Bœsch
parent 241f8465d0
commit aecf0cf5ed
2 changed files with 246 additions and 1 deletions

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@ -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))

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@ -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);
}