我在使用ffmpeg解码wav文件时遇到问题。我是新手，我不太习惯它。
在我的应用程序中，我必须输入音频文件并获取一系列样本。 我使用ffmpeg来创建一个函数，它输入文件的路径，开始输出样本的时间位置以及以秒为单位解码的块的长度。
我没有声誉，所以我必须制作一个gdrive目录，在那里你可以看到问题和我工作的文件。
当我尝试解码文件 harp.wav 时，一切运行正常，我可以将图像绘制成图像 plot-harp.png
该文件是一个WAV文件，编码为：pcm_u8,11025 Hz，1个通道，u8,88 kb / s
当我尝试解码文件 demo-unprocessed.wav 时出现问题。 它输出一系列无意义的样本。它会输出一系列样本，如图像 graph1-demo.jpg 所示。
该文件是一个WAV文件，编码为：pcm_s16le，44100 Hz，1个通道，s16,705 kb / s
我的代码中存在问题的IDK，我已经在使用FFMPEG解码之前和之后检查了代码，并且它工作得很好。
以下是dataReader.cpp的代码：
/* Start by including the necessary */
#include "dataReader.h"
#include <cstdlib>
#include <iostream>
#include <fstream>
#ifdef __cplusplus
extern "C" {
#endif
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#ifdef __cplusplus
}
#endif
using namespace std;
/* initialization function for audioChunk */
audioChunk::audioChunk(){
data=NULL;
size=0;
bitrate=0;
}
/* function to get back chunk lenght in seconds */
int audioChunk::getTimeLenght(){
return size/bitrate;
}
/* initialization function for audioChunk_dNorm */
audioChunk_dNorm::audioChunk_dNorm(){
data=NULL;
size=0;
bitrate=0;
}
/* function to get back chunk lenght in seconds */
int audioChunk_dNorm::getTimeLenght(){
return size/bitrate;
}
/* function to normalize audioChunk into audioChunk_dNorm */
void audioChunk_dNorm::fillAudioChunk(audioChunk* cnk){
size=cnk->size;
bitrate=cnk->bitrate;
double min=cnk->data[0];
double max=cnk->data[0];
for(int i=0;i<cnk->size;i++){
if(*(cnk->data+i)>max) max=*(cnk->data+i);
else if(*(cnk->data+i)<min) min=*(cnk->data+i);
}
data=new double[size];
for(int i=0;i<size;i++){
//data[i]=cnk->data[i]+256*data[i+1];
if(data[i]!=255) data[i]=2*((cnk->data[i])-(max-min)/2)/(max-min);
else data[i]=0;
}
cout<<"bitrate "<<bitrate<<endl;
}
audioChunk readData(const char* path_name, const double start_time, const double lenght){
/* inizialize audioChunk */
audioChunk output;
/* Check input times */
if((start_time<0)||(lenght<0)) {
cout<<"Input times should be positive";
return output;
}
/* Start FFmpeg */
av_register_all();
/* Initialize the frame to read the data and verify memory allocation */
AVFrame* frame = av_frame_alloc();
if (!frame)
{
cout << "Error allocating the frame" << endl;
return output;
}
/* Initialization of the Context, to open the file */
AVFormatContext* formatContext = NULL;
/* Opening the file, and check if it has opened */
if (avformat_open_input(&formatContext, path_name, NULL, NULL) != 0)
{
av_frame_free(&frame);
cout << "Error opening the file" << endl;
return output;
}
/* Find the stream info, if not found, exit */
if (avformat_find_stream_info(formatContext, NULL) < 0)
{
av_frame_free(&frame);
avformat_close_input(&formatContext);
cout << "Error finding the stream info" << endl;
return output;
}
/* Check inputs to verify time input */
if(start_time>(formatContext->duration/1000000)){
cout<< "Error, start_time is over file duration"<<endl;
av_frame_free(&frame);
avformat_close_input(&formatContext);
return output;
}
/* Chunk = number of samples to output */
long long int chunk = ((formatContext->bit_rate)*lenght/8);
/* Start = address of sample where start to read */
long long int start = ((formatContext->bit_rate)*start_time/8);
/* Tot_sampl = number of the samples in the file */
long long int tot_sampl = (formatContext->bit_rate)*(formatContext->duration)/8000000;
/* Set the lenght of chunk to avoid segfault and to read all the file */
if (start+chunk>tot_sampl) {chunk = tot_sampl-start;}
if (lenght==0) {start = 0; chunk = tot_sampl;}
/* initialize the array to output */
output.data = new unsigned char[chunk];
output.bitrate = formatContext->bit_rate;
output.size=chunk;
av_dump_format(formatContext,0,NULL,0);
cout<<chunk<<" n of sample to read"<<endl;
cout<<start<<" start"<<endl;
cout<<output.bitrate<<" bitrate"<<endl;
cout<<tot_sampl<<" total sample"<<endl;
/* Find the audio Stream, if no audio stream are found, clean and exit */
AVCodec* cdc = NULL;
int streamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &cdc, 0);
if (streamIndex < 0)
{
av_frame_free(&frame);
avformat_close_input(&formatContext);
cout << "Could not find any audio stream in the file" << endl;
return output;
}
/* Open the audio stream to read data in audioStream */
AVStream* audioStream = formatContext->streams[streamIndex];
/* Initialize the codec context */
AVCodecContext* codecContext = audioStream->codec;
codecContext->codec = cdc;
/* Open the codec, and verify if it has opened */
if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
{
av_frame_free(&frame);
avformat_close_input(&formatContext);
cout << "Couldn't open the context with the decoder" << endl;
return output;
}
/* Initialize buffer to store compressed packets */
AVPacket readingPacket;
av_init_packet(&readingPacket);
int j=0;
int count = 0;
while(av_read_frame(formatContext, &readingPacket)==0){
if((count+readingPacket.size)>start){
if(readingPacket.stream_index == audioStream->index){
AVPacket decodingPacket = readingPacket;
// Audio packets can have multiple audio frames in a single packet
while (decodingPacket.size > 0){
// Try to decode the packet into a frame
// Some frames rely on multiple packets, so we have to make sure the frame is finished before
// we can use it
int gotFrame = 0;
int result = avcodec_decode_audio4(codecContext, frame, &gotFrame, &decodingPacket);
count += result;
if (result >= 0 && gotFrame)
{
decodingPacket.size -= result;
decodingPacket.data += result;
int a;
for(int i=0;i<result-1;i++){
*(output.data+j)=frame->data[0][i];
j++;
if(j>=chunk) break;
}
// We now have a fully decoded audio frame
}
else
{
decodingPacket.size = 0;
decodingPacket.data = NULL;
}
if(j>=chunk) break;
}
}
}else count+=readingPacket.size;
// To prevent memory leak, must free packet.
av_free_packet(&readingPacket);
if(j>=chunk) break;
}
// Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
// is set, there can be buffered up frames that need to be flushed, so we'll do that
if (codecContext->codec->capabilities & CODEC_CAP_DELAY)
{
av_init_packet(&readingPacket);
// Decode all the remaining frames in the buffer, until the end is reached
int gotFrame = 0;
int a;
int result=avcodec_decode_audio4(codecContext, frame, &gotFrame, &readingPacket);
while (result >= 0 && gotFrame)
{
// We now have a fully decoded audio frame
for(int i=0;i<result-1;i++){
*(output.data+j)=frame->data[0][i];
j++;
if(j>=chunk) break;
}
if(j>=chunk) break;
}
}
// Clean up!
av_free(frame);
avcodec_close(codecContext);
avformat_close_input(&formatContext);
cout<<"Ended Reading, "<<j<<" samples read"<<endl;
output.size=j;
return output;
}
这是dataReader.h
/*
* File: dataReader.h
* Author: davide
*
* Created on 27 luglio 2015, 11.11
*/
#ifndef DATAREADER_H
#define DATAREADER_H
/* function that reads a file and outputs an array of samples
* @ path_name = the path of the file to read
* @ start_time = the position where to start the data reading, 0 = start
* the time is in seconds, it can hold to 10e-6 seconds
* @ lenght = the lenght of the frame to extract the data,
* 0 = read all the file (do not use with big files)
* if lenght > of file duration, it reads through the end of file.
* the time is in seconds, it can hold to 10e-6 seconds
*/
#include <stdint.h>
class audioChunk{
public:
uint8_t *data;
unsigned int size;
int bitrate;
int getTimeLenght();
audioChunk();
};
class audioChunk_dNorm{
public:
double* data;
unsigned int size;
int bitrate;
int getTimeLenght();
void fillAudioChunk(audioChunk* cnk);
audioChunk_dNorm();
};
audioChunk readData(const char* path_name, const double start_time, const double lenght);
#endif /* DATAREADER_H */
最后还有应用程序的main.cpp。
/*
* File: main.cpp
* Author: davide
*
* Created on 28 luglio 2015, 17.04
*/
#include <cstdlib>
#include "dataReader.h"
#include "transforms.h"
#include "tognuplot.h"
#include <fstream>
#include <iostream>
using namespace std;
/*
*
*/
int main(int argc, char** argv) {
audioChunk *chunk1=new audioChunk;
audioChunk_dNorm *normChunk1=new audioChunk_dNorm;
*chunk1=readData("./audio/demo-unprocessed.wav",0,1);
normChunk1->fillAudioChunk(chunk1);
ofstream file1;
file1.open("./file/2wave.txt", std::ofstream::trunc);
if(file1.is_open()) {
for(int i=0;i<chunk1->size;i++) {
int a=chunk1->data[i];
file1<<i<<" "<<a<<endl;
}
}
else cout<<"Error opening file";
file1.close();
return 0;
}
我无法理解为什么输出会像这样。是否有可能解码器无法将样本（pcm_16le，16bits）转换为存储样本ad uint8_t的FFMPEG AVFrame.data？如果它有一些方法可以使FFMPEG适用于存储样本超过8位的音频文件？
文件graph1-demo_good.jpg是样本的样本，使用我制作的LIBSNDFILE应用程序提取。
编辑：好像程序无法将解码后的数据，存储在几个uint8_t unsigned char中的小端字节耦合到目标格式（我设置为unsigned char [ ]），因为它将位存储为小端16字节。因此，将数据导入audioChunk.data是正确的，但我必须将其读作无符号字符，而是作为几个小端字节。
答案 0 :(得分：0)
我使用gdb查看chunk1->data指向的内存。 （x /256xh 0x18dddf0，以十六进制转储前256个半字）。它看起来像有符号的16位值，因为它开始时有很多0，0xFFFF和0x0001。
所以你的代码需要让ffmpeg转换为特定的格式。 IDK如何最好地做到这一点，抱歉。