最近開始研究 ffmpeg，關注一下博主。

FFmpeg有個恥辱名單，LZ聽所過嗎(⊙_⊙)？

和我的程序有關系嗎？況且我的這個程序附帶了源碼，雖然簡單。

不錯,收藏,學習

找到不能播放aac和ogg的問題了，原因是在ffmpeg里分配內存需要用av_malloc，釋放內存要用av_free，因為windows和linux下內存分配存在不同，而ffmpeg在解碼的時候是要檢查內存是否對齊（內存對齊可以加快CPU的處理速度），所以在程序里在window環境下通過malloc或者通過數組的方式分配的內存不完全是內存對齊的，所以在遇到aac和ogg這種幀長度與其它音頻格式幀長度不一致時，就有可能在運行時出錯。修改后的代碼如下，讀者參照著自己修改即可。

#include <stdio.h>
#include <stdlib.h>
#include <windows.h>
#include <mmsystem.h>

#pragma comment(lib, "winmm.lib")

#ifdef __cplusplus
extern "C" {
#endif

#include "./include/avcodec.h"
#include "./include/avformat.h"
#include "./include/avutil.h"
#include "./include/mem.h"

#ifdef __cplusplus
}
#endif

#define BLOCK_SIZE 4608
#define BLOCK_COUNT 20

HWAVEOUT hWaveOut = NULL;

static void CALLBACK waveOutProc(HWAVEOUT, UINT, DWORD, DWORD, DWORD);
static WAVEHDR* allocateBlocks(int size, int count);
static void freeBlocks(WAVEHDR* blockArray);
static void writeAudio(HWAVEOUT hWaveOut, LPSTR data, int size);

static CRITICAL_SECTION waveCriticalSection;
static WAVEHDR* waveBlocks;
static volatile unsigned int waveFreeBlockCount;
static int waveCurrentBlock;

typedef struct AudioState {
AVFormatContext* pFmtCtx;
AVCodecContext* pCodecCtx;
AVCodec* pCodec;
//uint8_t* audio_buf1[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
uint8_t* audio_buf1;
uint8_t* audio_buf;
unsigned int audio_buf_size; /* in bytes */
unsigned int buffer_size;
int audio_buf_index; /* in bytes */
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
uint8_t* audio_pkt_data;
int audio_pkt_size;

int stream_index;
} AudioState;

int audio_decode_frame(AudioState* pState) {

AVPacket* pkt_temp = &pState->audio_pkt_temp;
AVPacket* pkt = &pState->audio_pkt;
AVCodecContext* dec= pState->pCodecCtx;
int len = 0, data_size = sizeof(pState->audio_buf1);
int err = 0;

for( ; ; ) {
while (pkt_temp->size > 0) {
// data_size = sizeof(pState->audio_buf1);
data_size = pState->buffer_size;
len = avcodec_decode_audio3(dec, (int16_t*)pState->audio_buf1, &data_size, pkt_temp);
if (len < 0) {
pkt_temp->size = 0;
break;
}

pkt_temp->data += len;
pkt_temp->size -= len;

if (data_size <= 0)
continue;

pState->audio_buf = pState->audio_buf1;
return data_size;
}

if (pkt->data)
av_free_packet(pkt);

if((err = av_read_frame(pState->pFmtCtx, pkt)) < 0)
return -1;

pkt_temp->data = pkt->data;
pkt_temp->size = pkt->size;
}

return -1;
}

int main(int argc, char* argv[]) {
int err = 0;
AudioState audio_state = {0};
unsigned int i = 0;
unsigned int ready = 0;

OPENFILENAME ofn = {0};
char filename[MAX_PATH];
WAVEFORMATEX wfx = {0};
uint8_t buffer[BLOCK_SIZE];
uint8_t* pbuffer = buffer;
AVInputFormat* iformat = NULL;

int audio_size = 0, data_size = 0;
int len = 0, len1 = 0, eof = 0, size = 0;

memset(&ofn, 0, sizeof(OPENFILENAME));
ofn.lStructSize = sizeof(ofn);
ofn.hwndOwner = GetDesktopWindow();
ofn.lpstrFile = filename;
ofn.lpstrFile[0] = '\0';
ofn.nMaxFile = sizeof(filename) / sizeof(filename[0]);
ofn.lpstrFilter = _TEXT("All support files\0*.aac;*.ape;*.flac;*.mp3;*.mp4;*.mpc;*.ogg;*.tta;*.wma;*.wav\0");
ofn.nFilterIndex = 1;
ofn.lpstrFileTitle = NULL;
ofn.nMaxFileTitle = 0;
ofn.lpstrInitialDir = NULL;
ofn.Flags = OFN_PATHMUSTEXIST | OFN_FILEMUSTEXIST;

if (GetOpenFileName(&ofn) == FALSE)
return 0;

av_register_all();

err = av_open_input_file(&audio_state.pFmtCtx, filename, NULL, 0, NULL);
if(err < 0) {
printf("can not open file %s.\n", filename);
return -1;
}

err = av_find_stream_info(audio_state.pFmtCtx);
if(err < 0) {
printf("can not find stream info of file %s.\n", filename);
return -1;
}

for(i = 0; i < audio_state.pFmtCtx->nb_streams; i++) {
if(audio_state.pFmtCtx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) {
audio_state.pCodecCtx = audio_state.pFmtCtx->streams[i]->codec;
audio_state.stream_index = i;
ready = 1;
break;
}
}

if(!ready)
return -1;

audio_state.pCodec = avcodec_find_decoder(audio_state.pCodecCtx->codec_id);
if(!audio_state.pCodec || avcodec_open(audio_state.pCodecCtx, audio_state.pCodec) < 0)
return -1;

wfx.nSamplesPerSec = audio_state.pCodecCtx->sample_rate;
switch(audio_state.pCodecCtx->sample_fmt)
{
case SAMPLE_FMT_U8:
wfx.wBitsPerSample = 8;
break;
case SAMPLE_FMT_S16:
wfx.wBitsPerSample = 16;
break;
case SAMPLE_FMT_S32:
wfx.wBitsPerSample = 32;
break;
case SAMPLE_FMT_FLT:
wfx.wBitsPerSample = sizeof(double) * 8;
break;
default:
wfx.wBitsPerSample = 0;
break;
}

wfx.nChannels = FFMIN(2, audio_state.pCodecCtx->channels);
wfx.cbSize = 0;
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nBlockAlign = (wfx.wBitsPerSample * wfx.nChannels) >> 3;
wfx.nAvgBytesPerSec = wfx.nBlockAlign * wfx.nSamplesPerSec;

waveBlocks = allocateBlocks(BLOCK_SIZE, BLOCK_COUNT);
waveFreeBlockCount = BLOCK_COUNT;
waveCurrentBlock = 0;

InitializeCriticalSection(&waveCriticalSection);

// open wave out device
if(waveOutOpen(&hWaveOut, WAVE_MAPPER, &wfx, (DWORD_PTR)waveOutProc,
(DWORD_PTR)&waveFreeBlockCount, CALLBACK_FUNCTION) != MMSYSERR_NOERROR) {
fprintf(stderr, "%s: unable to open wave mapper device\n", argv[0]);
ExitProcess(1);
}

// allocate memory
audio_state.audio_buf1 =(uint8_t*)av_malloc(buffer_size);
audio_state.buffer_size = buffer_size;

// play loop
for( ; ; ) {

len = BLOCK_SIZE;
size = 0;
pbuffer = buffer;

if(eof)
break;

while(len > 0) {
if(audio_state.audio_buf_index >= (int)audio_state.audio_buf_size) {
audio_size = audio_decode_frame(&audio_state);
if(audio_size < 0) {
if(size > 0)
break;

eof = 1;
break;
}

audio_state.audio_buf_size = audio_size;
audio_state.audio_buf_index = 0;
}

len1 = audio_state.audio_buf_size - audio_state.audio_buf_index;
if(len1 > len)
len1 = len;

memcpy(pbuffer, (uint8_t *)audio_state.audio_buf + audio_state.audio_buf_index, len1);

len -= len1;
pbuffer += len1;
size += len1;
audio_state.audio_buf_index += len1;
}

writeAudio(hWaveOut, (char*)buffer, size);
}

// free allocated memory
av_free(audio_state.audio_buf1);
audio_state.audio_buf1 = NULL;

// wait for complete
for( ; ; ) {
if(waveFreeBlockCount >= BLOCK_COUNT)
break;

Sleep(10);
}

for(i = 0; i < waveFreeBlockCount; i++)
if(waveBlocks[i].dwFlags & WHDR_PREPARED)
waveOutUnprepareHeader(hWaveOut, &waveBlocks[i], sizeof(WAVEHDR));

DeleteCriticalSection(&waveCriticalSection);
freeBlocks(waveBlocks);
waveOutClose(hWaveOut);

avcodec_close(audio_state.pCodecCtx);

system("pause");
return 0;
}

static void writeAudio(HWAVEOUT hWaveOut, LPSTR data, int size)
{
WAVEHDR* current;
int remain;

current = &waveBlocks[waveCurrentBlock];

while(size > 0) {
/*
* first make sure the header we're going to use is unprepared
*/
if(current->dwFlags & WHDR_PREPARED)
waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));

if(size < (int)(BLOCK_SIZE - current->dwUser)) {
memcpy(current->lpData + current->dwUser, data, size);
current->dwUser += size;
break;
}

remain = BLOCK_SIZE - current->dwUser;
memcpy(current->lpData + current->dwUser, data, remain);
size -= remain;
data += remain;
current->dwBufferLength = BLOCK_SIZE;

waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));

EnterCriticalSection(&waveCriticalSection);
waveFreeBlockCount--;
LeaveCriticalSection(&waveCriticalSection);

/*
* wait for a block to become free
*/
while(!waveFreeBlockCount)
Sleep(10);

/*
* point to the next block
*/
waveCurrentBlock++;
waveCurrentBlock %= BLOCK_COUNT;

current = &waveBlocks[waveCurrentBlock];
current->dwUser = 0;
}
}

static WAVEHDR* allocateBlocks(int size, int count)
{
char* buffer;
int i;
WAVEHDR* blocks;
DWORD totalBufferSize = (size + sizeof(WAVEHDR)) * count;

/*
* allocate memory for the entire set in one go
*/
if((buffer = (char*)HeapAlloc(
GetProcessHeap(),
HEAP_ZERO_MEMORY,
totalBufferSize
)) == NULL) {
fprintf(stderr, "Memory allocation error\n");
ExitProcess(1);
}

/*
* and set up the pointers to each bit
*/
blocks = (WAVEHDR*)buffer;
buffer += sizeof(WAVEHDR) * count;
for(i = 0; i < count; i++) {
blocks[i].dwBufferLength = size;
blocks[i].lpData = buffer;
buffer += size;
}

return blocks;
}

static void freeBlocks(WAVEHDR* blockArray)
{
/*
* and this is why allocateBlocks works the way it does
*/
HeapFree(GetProcessHeap(), 0, blockArray);
}

static void CALLBACK waveOutProc(
HWAVEOUT hWaveOut,
UINT uMsg,
DWORD dwInstance,
DWORD dwParam1,
DWORD dwParam2
)
{
int* freeBlockCounter = (int*)dwInstance;
/*
* ignore calls that occur due to opening and closing the
* device.
*/
if(uMsg != WOM_DONE)
return;

EnterCriticalSection(&waveCriticalSection);
(*freeBlockCounter)++;
LeaveCriticalSection(&waveCriticalSection);
}

你的下載包里面,沒有.avcodec-52.dll 所以無法運行.

當初我也做了一個MSYS+MinGW，想研究一下ffmpeg.發現MSYS本質不是應用于linux的，如果做windows開發，用ffmpeg sdk好像足夠了。
偶沒有時間做了，那位大蝦搭建了cygwin+MinGW，希望共同研究一下ffmpeg在linux下特別時嵌入式下的應用。

謝謝你的代碼。問題已經解決了。

感謝樓主的代碼。把av_read_frame那兩行改成下面這樣，就可以直接播絕大部分視頻文件了（聲音部分）。

pkt->stream_index = -1;
while(pkt->stream_index != pState->stream_index) {
if((err = av_read_frame(pState->pFmtCtx, pkt)) < 0)
return -1;
}

暴風是不是你弄的

@glueless lace wigs
不是，要是有那本事，就不玩了。

謝謝jackzz，在我眼里你已經很牛了，剛剛在WinCE上用你的修正后代碼測試了一下mp3、flac、ape、ogg、wma、wav，都播得很happy，不錯不錯。

我的QQ是475316440~麻煩加一下，關于解碼，我有一些疑問~thanks~
簡述就是avcodec_decode_audio3((AVCodecContext *avctx, int16_t *samples,int *frame_size_ptr,AVPacket *avpkt))解出一幀的數據，這寫數據，不總是在samples的最前面順序排列？