title: ffmpeg_sample解读_filter_audio
date: 2020-10-28 10:15:02
tags: [读书笔记]
typora-copy-images-to: ./imgs
typora-root-url: ./imgs
总结
本示例将生成一个正弦的音频PCM数据,然后把PCM数据经过如下filterchain的处理。把输出的每一帧PCM数据的MD5值打印出来.
这里就涉及了过滤器相关的
流程图
graph TB
afa[av_frame_alloc]
-->ama[av_md5_alloc]
-->ifg[init_filter_graph]
-->afga[avfilter_graph_alloc]
-->afgbn[avfilter_get_by_name]
-->afaf[avfilter_graph_alloc_filter]
-->afis[avfilter_init_str]
-->afl[avfilter_link]
-->afgc[avfilter_graph_config]
-->gi[get_input]
-->abaf[av_buffersrc_add_frame]
-->abgf[av_buffersink_get_frame]
-->free[free_all]
其实相对好理解. 这里创建了拿到四个过滤器,创建过滤器上下文,设置参数. 用三种方式是实现,然后把四个过滤器连接起来,最后把随机生成的帧,送入第一个过滤器.然后从最后一个过滤器中取出数据.打印md5
代码
/**
* @file
* libavfilter API usage example.
*
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
*
* abuffer: This provides the endpoint where you can feed the decoded samples.
* volume: In this example we hardcode it to 0.90.
* aformat: This converts the samples to the samplefreq, channel layout,
* and sample format required by the audio device.
* abuffersink: This provides the endpoint where you can read the samples after
* they have passed through the filter chain.
*/
#include
#include
#include
#include
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
/**
* 创建了四个过滤器上下文.初始化参数,然后连接到一起
* 初始化过滤器. 这里应该是各种转换都是不同的过滤器效果.一层一层处理
* @param graph
* @param src
* @param sink
* @return
* 过滤器AVFilter和过滤器上下文AVFilterContext的关系
* 过滤器是sdk内置的,我们直接可以使用.而过滤器上下文则是相关的环境和数据.
* 我们总是要根据过滤器来初始化过滤器上下文. 上下文是和接受的数据相关的.我们操作的数据也是操作上下文
*/
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
AVFilterContext **sink) {
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
const AVFilter *abuffer;
AVFilterContext *volume_ctx;
const AVFilter *volume;
AVFilterContext *aformat_ctx;
const AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
const AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
uint8_t ch_layout[64];
int err;
/* Create a new filtergraph, which will contain all the filters. */
filter_graph = avfilter_graph_alloc();//分配过滤器图形控件
if (!filter_graph) {
fprintf(stderr, "Unable to create filter graph.\n");
return AVERROR(ENOMEM);
}
/* Create the abuffer filter;
* it will be used for feeding the data into the graph. */
//------1------通过名称初始化 abuffer过滤器. 看起来应该是用来填充数据用的
abuffer = avfilter_get_by_name("abuffer");
if (!abuffer) {
fprintf(stderr, "Could not find the abuffer filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
//通过已有的filter_graph 过滤器图形生成,名称是src,返回过滤器上下文
abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
if (!abuffer_ctx) {
fprintf(stderr, "Could not allocate the abuffer instance.\n");
return AVERROR(ENOMEM);
}
//给上边的 过滤器上下文abuffer_ctx设置参数. 总共设置了 四个参数.音频相关 time_base和sample_rate 是互为倒数
/* Set the filter options through the AVOptions API. */
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set(abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set(abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT),
AV_OPT_SEARCH_CHILDREN);
av_opt_set_q(abuffer_ctx, "time_base", (AVRational) {1, INPUT_SAMPLERATE},
AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
/* Now initialize the filter; we pass NULL options, since we have already
* set all the options above. */
//又根据上表初始化完成的 过滤器上下文来初始化过滤器
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffer filter.\n");
return err;
}
//------2------获取音量过滤器,然后创建音量过滤器上下文,在给他设置参数,和上边的过滤器类似
/* Create volume filter. */
volume = avfilter_get_by_name("volume");
if (!volume) {
fprintf(stderr, "Could not find the volume filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
//初始化一个音量过滤器上下文,名称是volume
volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
if (!volume_ctx) {
fprintf(stderr, "Could not allocate the volume instance.\n");
return AVERROR(ENOMEM);
}
/* A different way of passing the options is as key/value pairs in a
* dictionary. */
//-另一种给上下文设置参数的方式.总之是先把参数设置给AVDictionary自带,在设置给上下文
av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
err = avfilter_init_dict(volume_ctx, &options_dict);
av_dict_free(&options_dict);
if (err < 0) {
fprintf(stderr, "Could not initialize the volume filter.\n");
return err;
}
/* Create the aformat filter;
* it ensures that the output is of the format we want. */
//------3------在找到格式过滤器
aformat = avfilter_get_by_name("aformat");
if (!aformat) {
fprintf(stderr, "Could not find the aformat filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
//初始化格式过滤器上下文,命名为aformat
aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
if (!aformat_ctx) {
fprintf(stderr, "Could not allocate the aformat instance.\n");
return AVERROR(ENOMEM);
}
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
//第三种方式,把key,value写入 options_str这个buf中, 其实就是给options_str设置值
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t) AV_CH_LAYOUT_STEREO);
//用上边设置的str 初始化过滤器上下文
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
return err;
}
/* Finally create the abuffersink filter;
* it will be used to get the filtered data out of the graph. */
//------4------在找到一个过滤器,下边肯定会又初始化一个上下文,设置参数. 方式都是一样的
abuffersink = avfilter_get_by_name("abuffersink");
if (!abuffersink) {
fprintf(stderr, "Could not find the abuffersink filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
//找到过滤器上下文
abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
if (!abuffersink_ctx) {
fprintf(stderr, "Could not allocate the abuffersink instance.\n");
return AVERROR(ENOMEM);
}
/* This filter takes no options. */
//没有默认参数来初始化上下文
err = avfilter_init_str(abuffersink_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffersink instance.\n");
return err;
}
/* Connect the filters;
* in this simple case the filters just form a linear chain. */
//把四个过滤器上下文进行连接,形成一个单链,上个过滤器的输出就是下个过滤器的输入
err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
if (err >= 0)
err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
if (err >= 0)
err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
if (err < 0) {
fprintf(stderr, "Error connecting filters\n");
return err;
}
/* Configure the graph. */ //配置过滤器图形.我理解,四个过滤器上下文都是通过过滤器图形来创建的.他们肯定是包含在内
err = avfilter_graph_config(filter_graph, NULL);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
return err;
}
*graph = filter_graph;
//这两个分别是输入数据和输出数据的过滤器上下文
*src = abuffer_ctx;
*sink = abuffersink_ctx;
return 0;
}
/* Do something useful with the filtered data: this simple
* example just prints the MD5 checksum of each plane to stdout. */
static int process_output(struct AVMD5 *md5, AVFrame *frame) {
int planar = av_sample_fmt_is_planar(frame->format);
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
int i, j;
for (i = 0; i < planes; i++) {
uint8_t checksum[16];
av_md5_init(md5);
av_md5_sum(checksum, frame->extended_data[i], plane_size);
fprintf(stdout, "plane %d: 0x", i);
for (j = 0; j < sizeof(checksum); j++)
fprintf(stdout, "%02X", checksum[j]);
fprintf(stdout, "\n");
}
fprintf(stdout, "\n");
return 0;
}
/* Construct a frame of audio data to be filtered;
* this simple example just synthesizes a sine wave. */
//产生一帧数据.随机生成的
static int get_input(AVFrame *frame, int frame_num) {
int err, i, j;
#define FRAME_SIZE 1024
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
err = av_frame_get_buffer(frame, 0);
if (err < 0)
return err;
/* Fill the data for each channel. */
for (i = 0; i < 5; i++) {
float *data = (float *) frame->extended_data[i];
for (j = 0; j < frame->nb_samples; j++)
data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
}
return 0;
}
/**
* 本示例将生成一个正弦的音频PCM数据,然后把PCM数据经过如下filterchain的处理。把输出的每一帧PCM数据的MD5值打印出来
* @param argc
* @param argv
* @return
* avfilter_graph_alloc_filter: 在filtergraph中创建一个filter实例。
avfilter_init_str:使用提供的字符串参数初始化一个filter。
avfilter_init_dict:使用提供的AVDictionary初始化一个filter。
avfilter_link:把两个filter连接在一起。
作者:smallest_one
链接:https://www.jianshu.com/p/f677992bbde9
来源:
著作权归作者所有。商业转载请联系作者获得授权,非商业转载请注明出处。
*/
int filter_audio_main(int argc, char *argv[]) {
struct AVMD5 *md5;
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
if (argc < 2) {
fprintf(stderr, "Usage: %s \n", argv[0]);
return 1;
}
duration = atof(argv[1]);//char 转float
nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;//帧数
if (nb_frames <= 0) {
fprintf(stderr, "Invalid duration: %s\n", argv[1]);
return 1;
}
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc(); //初始化未压缩的帧
if (!frame) {
fprintf(stderr, "Error allocating the frame\n");
return 1;
}
md5 = av_md5_alloc();//初始化一个AVMD5
if (!md5) {
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
/* Set up the filtergraph. */
//初始化过滤器视图,src是最初的过滤器上下文.负责输入数据 sink是最后的过滤器上下文.负责输出数据
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
fprintf(stderr, "Unable to init filter graph:");
goto fail;
}
//一帧一帧的循环遍历
/* the main filtering loop */
for (i = 0; i < nb_frames; i++) {
/* get an input frame to be filtered */
err = get_input(frame, i);
if (err < 0) {
fprintf(stderr, "Error generating input frame:");
goto fail;
}
/* Send the frame to the input of the filtergraph. */
//把帧数据放到 过滤器中处理
err = av_buffersrc_add_frame(src, frame);
if (err < 0) {
av_frame_unref(frame);
fprintf(stderr, "Error submitting the frame to the filtergraph:");
goto fail;
}
//从最后的过滤器中把数据取出.放回到frame中,这里也就是过滤器完成了帧的处理
/* Get all the filtered output that is available. */
while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
/* now do something with our filtered frame */
// 打印处理完的帧的md5
err = process_output(md5, frame);
if (err < 0) {
fprintf(stderr, "Error processing the filtered frame:");
goto fail;
}
av_frame_unref(frame);
}
if (err == AVERROR(EAGAIN)) {
/* Need to feed more frames in. */
continue;
} else if (err == AVERROR_EOF) {
/* Nothing more to do, finish. */
break;
} else if (err < 0) {
/* An error occurred. */
fprintf(stderr, "Error filtering the data:");
goto fail;
}
}
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
return 0;
fail:
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}