我在使用 ffmpeg 解码 wav 文件时遇到问题。我是新手，不太习惯。
在我的应用程序中，我必须输入音频文件并获取要处理的样本数组。 我使用 ffmpeg 创建了一个函数，用于输入文件的路径、开始输出样本的时间位置以及要在几秒钟内解码的 block 的长度。
我没有声誉，所以我不得不创建一个 gdrive 目录，您可以在其中查看问题和我处理的文件。
当我尝试解码文件 harp.wav 时，一切运行正常，我可以绘制样本，如图像 plot-harp.png
该文件是一个 WAV 文件，编码为:pcm_u8, 11025 Hz, 1 channels, u8, 88 kb/s
当我尝试解码文件 demo-unprocessed.wav 时，问题就来了。 它输出一系列没有意义的样本。它输出一系列样本，如图 graph1-demo.jpg 所示。
该文件是一个 WAV 文件，编码为:pcm_s16le, 44100 Hz, 1 channels, s16, 705 kb/s
IDK 我的代码中的问题所在，我已经用 FFMPEG 解码前后检查了代码，它工作得非常好。
这是 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;
}
这里是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 */
最后是应用程序的 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;
}
我不明白为什么输出是这样的。解码器是否有可能无法将样本(pcm_16le，16 位)转换为 FFMPEG AVFrame.data，它存储样本 ad uint8_t？如果是，是否有某种方法可以使 FFMPEG 用于存储超过 8 位样本的音频文件？
文件 graph1-demo_good.jpg 是示例的样子，使用我制作的工作 LIBSNDFILE 应用程序提取。
编辑: 似乎程序无法将解码数据(存储在几个 uint8_t unsigned char 中的几个小字节序字节)转换为目标格式(我设置为 unsigned char[ ]), 因为它将位存储为 little-endian 16 字节。所以 audioChunk.data 中的数据是正确的，但我必须不是将其作为 unsigned char 来读取，而是作为几个 little-endian 字节来读取。
最佳答案
我用gdb查看了chunk1->data指向的内存。 (x/256xh 0x18dddf0，以十六进制转储前 256 个半字)。它看起来像是带符号的 16 位值，因为它以大量 0、0xFFFF 和 0x0001 开头。
因此您的代码需要请求 ffmpeg 转换为特定格式。 IDK 如何最好地做到这一点，抱歉。
关于c++ - 用 FFMPEG 解码 pcm_s16le？，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/31730161/