#Filter和SDL（Audio）
本文主要来自官方例子doc/examples/filtering_video.c 。
- 滤镜官方语法 , 推荐参考《FFmpeg从入门到精通》。
##使用滤镜流程 参考上一篇视频滤镜使用流程 。注意以下一点：
- 获取滤镜器的名称
输入：avfilter_get_by_name("buffer") -> avfilter_get_by_name("abuffer")
输出：avfilter_get_by_name("buffersink") -> avfilter_get_by_name("abuffersink")
其中，AVFormatContext、AVPacket等重要的结构体请看：FFmpeg重要结构体 。
##代码实现
/**
* @author 秦城季
* @email xhunmon@126.com
* @Blog https://qincji.gitee.io
* @date 2021/01/10
* description: 来自官方例子：doc/examples/filtering_audio.c
* <br>
*/
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <SDL.h>
};
//fltp
//static const char *filter_descr = "aresample=44100,aformat=sample_fmts=fltp:channel_layouts=mono";
static const char *filter_descr = "aecho=0.8:0.88:60:0.4";//参考：http://ffmpeg.org/ffmpeg-filters.html#aecho
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_FLTP, AV_SAMPLE_FMT_NONE};
static const int64_t out_channel_layouts[] = {AV_CH_LAYOUT_STEREO, -1};
static const int out_sample_rates[] = {44100, -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%lld",
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(static_cast<AVSampleFormat>(outlink->format)), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static Uint8 *audio_chunk;
static Uint32 audio_len;
static Uint8 *audio_pos;
void fill_audio(void *udata, Uint8 *stream, int len) {
//SDL 2.0
SDL_memset(stream, 0, len);
if (audio_len == 0)
return;
len = (len > audio_len ? audio_len : len);
SDL_MixAudio(stream, audio_pos, len, SDL_MIX_MAXVOLUME);
audio_pos += len;
audio_len -= len;
}
//https://blog.csdn.net/leixiaohua1020/article/details/40544521
static int init_sdl(AVCodecContext *dec_ctx) {
int ret = -1;
// B1. 初始化SDL子系统：缺省(事件处理、文件IO、线程)、视频、音频、定时器
if (SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
printf("SDL_Init() failed: %s\n", SDL_GetError());
goto end;
}
//注意：这里设置的参数会算出 audio_chunk 所使用的长度
//audio_chunk = 采样数 * 通道数 * 位宽
SDL_AudioSpec wanted_spec;
wanted_spec.freq = dec_ctx->sample_rate;
// wanted_spec.format = dec_ctx->sample_fmt;
wanted_spec.format = AUDIO_F32SYS;//位宽=4
wanted_spec.channels = dec_ctx->channels;//通道数
wanted_spec.silence = 0;
wanted_spec.samples = dec_ctx->frame_size;//采样数
wanted_spec.callback = fill_audio;
if (SDL_OpenAudio(&wanted_spec, NULL) < 0) {
printf("can't open audio.\n");
goto end;
}
ret = 1;
//Play
SDL_PauseAudio(0);
end:
return ret;
}
static void sdl_play(const AVFrame *frame) {
if (frame->data[0][0] == '\0') {//没有数据？
return;
}
int i, ch, data_size;
data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);//每一个采样点所占的字节数
Uint32 len = data_size * frame->nb_samples * dec_ctx->channels;//所有通道采样数所占字节长度（一帧大小）
Uint8 *all_channels_buf = (Uint8 *) malloc(len);
int index = 0;
//把所有通道采样数据重新排列
for (i = 0; i < frame->nb_samples; i++) {
for (ch = 0; ch < dec_ctx->channels; ch++) {
memcpy(all_channels_buf + index * data_size, frame->data[ch] + data_size * i, data_size);
++index;
}
}
//把一帧数据设置给SDL播放器
audio_chunk = all_channels_buf;
audio_len = len;
audio_pos = audio_chunk;
while (audio_len > 0)//Wait until finish
SDL_Delay(1);
free(all_channels_buf);
}
static void print_frame(const AVFrame *frame) {
sdl_play(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);
}
const char *filename = "source/Kobe.flv";
if ((ret = open_input_file(filename)) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
if ((ret = init_sdl(dec_ctx)) < 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;
//切换查看与原来的效果差异
// print_frame(frame);
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);
}##测试文件下载地址
参考