来自雷神最简单的基于FFMPEG+SDL的音频播放器 ver2 （采用SDL2.0）
部分编码器（例如WMA）的AVCodecContext中的frame_size没有进行初始化,造成播放异常处理
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <iostream>
#define __STDC_CONSTANT_MACROS
#ifdef _WIN32
//Windows
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"
#include <libavutil/opt.h>
#include "SDL/SDL.h"
};
#else
//Linux...
#ifdef __cplusplus
extern "C"
{
#endif
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#include <SDL2/SDL.h>
#ifdef __cplusplus
};
#endif
#endif
#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
//Output PCM
#define OUTPUT_PCM 1
//Use SDL
#define USE_SDL 1
//Buffer:
//|-----------|-------------|
//chunk-------pos---len-----|
static Uint8 *audio_chunk;
static Uint32 audio_len;
static Uint8 *audio_pos;
/* The audio function callback takes the following parameters:
* stream: A pointer to the audio buffer to be filled
* len: The length (in bytes) of the audio buffer
*/
void fill_audio(void *udata, Uint8 *stream, int len) {
//SDL 2.0
SDL_memset(stream, 0, len);
if (audio_len <= 0)
return;
printf("audio_len:%d\n", audio_len);
len = (len>audio_len ? audio_len : len); /* Mix as much data as possible */
SDL_MixAudio(stream, audio_pos, len, SDL_MIX_MAXVOLUME);
audio_pos += len;
audio_len -= len;
}
//-----------------
/* Minimum SDL audio buffer size, in samples. */
#define SDL_AUDIO_MIN_BUFFER_SIZE 512
/* Calculate actual buffer size keeping in mind not cause too frequent audio callbacks */
#define SDL_AUDIO_MAX_CALLBACKS_PER_SEC 30
using namespace std;
int main(int argc, char* argv[])
{
AVFormatContext *pFormatCtx;
int i, audioStream;
AVCodecContext *pCodecCtx;
AVCodec *pCodec;
AVPacket *packet;
uint8_t *out_buffer;
AVFrame *pFrame;
SDL_AudioSpec wanted_spec;
int ret;
uint32_t len = 0;
int got_picture;
int index = 0;
int64_t in_channel_layout;
struct SwrContext *au_convert_ctx;
char url[] = "C:/Users/Public/Videos/Sample\ Videos/Wildlife.wmv";
av_register_all();
avformat_network_init();
pFormatCtx = avformat_alloc_context();
//Open
if (avformat_open_input(&pFormatCtx, url, NULL, NULL) != 0) {
printf("Couldn't open input stream.\n");
return -1;
}
// Retrieve stream information
if (avformat_find_stream_info(pFormatCtx, NULL)<0) {
printf("Couldn't find stream information.\n");
return -1;
}
// Dump valid information onto standard error
av_dump_format(pFormatCtx, 0, url, false);
// Find the first audio stream
audioStream = -1;
for (i = 0; i < pFormatCtx->nb_streams; i++)
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
audioStream = i;
break;
}
if (audioStream == -1) {
printf("Didn't find a audio stream.\n");
return -1;
}
// Get a pointer to the codec context for the audio stream
pCodecCtx = pFormatCtx->streams[audioStream]->codec;
// Find the decoder for the audio stream
pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
if (pCodec == NULL) {
printf("Codec not found.\n");
return -1;
}
// Open codec
if (avcodec_open2(pCodecCtx, pCodec, NULL)<0) {
printf("Could not open codec.\n");
return -1;
}
packet = (AVPacket *)av_malloc(sizeof(AVPacket));
av_init_packet(packet);
//Out Audio Param
uint64_t out_channel_layout = AV_CH_LAYOUT_STEREO;
//nb_samples: AAC-1024 MP3-1152
int out_nb_samples = pCodecCtx->frame_size;//可能为0
cout << "out_nb_samples:" << out_nb_samples << endl;
AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16;
int out_sample_rate = 44100;
int out_channels = av_get_channel_layout_nb_channels(out_channel_layout);
//Out Buffer Size
int out_buffer_size = av_samples_get_buffer_size(NULL, out_channels, out_nb_samples, out_sample_fmt, 1);
std::cout << "out bs:"<<out_buffer_size << std::endl;
out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE * 2);
pFrame = av_frame_alloc();
//SDL------------------
#if USE_SDL
//Init
if (SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) {
printf("Could not initialize SDL - %s\n", SDL_GetError());
return -1;
}
//SDL_AudioSpec
wanted_spec.freq = out_sample_rate;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = out_channels;
wanted_spec.silence = 0;
wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AUDIO_MAX_CALLBACKS_PER_SEC));;
wanted_spec.callback = fill_audio;
wanted_spec.userdata = pCodecCtx;
if (SDL_OpenAudio(&wanted_spec, NULL)<0) {
printf("can't open audio.\n");
return -1;
}
#endif
//FIX:Some Codec's Context Information is missing
in_channel_layout = av_get_default_channel_layout(pCodecCtx->channels);
//Swr
au_convert_ctx = swr_alloc();
au_convert_ctx = swr_alloc_set_opts(au_convert_ctx, out_channel_layout, out_sample_fmt, out_sample_rate,
in_channel_layout, pCodecCtx->sample_fmt, pCodecCtx->sample_rate, 0, NULL);
swr_init(au_convert_ctx);
//Play
SDL_PauseAudio(0);
while (av_read_frame(pFormatCtx, packet) >= 0) {
if (packet->stream_index == audioStream) {
ret = avcodec_decode_audio4(pCodecCtx, pFrame, &got_picture, packet);
if (ret < 0) {
printf("Error in decoding audio frame.\n");
return -1;
}
if (got_picture > 0) {
//重新获取buffer_size
out_buffer_size = av_samples_get_buffer_size(NULL, out_channels, pFrame->nb_samples, out_sample_fmt, 1);
swr_convert(au_convert_ctx, &out_buffer, MAX_AUDIO_FRAME_SIZE, (const uint8_t **)pFrame->data, pFrame->nb_samples);
#if 1
printf("index:%5d\t pts:%lld\t packet size:%d\n", index, packet->pts, packet->size);
#endif
index++;
}
#if USE_SDL
while (audio_len>0)//Wait until finish
SDL_Delay(1);
//Set audio buffer (PCM data)
audio_chunk = (Uint8 *)out_buffer;
printf("out_buffer_size:%d\n", out_buffer_size);
//Audio buffer length
audio_len = out_buffer_size;
audio_pos = audio_chunk;
#endif
}
av_free_packet(packet);
}
swr_free(&au_convert_ctx);
#if USE_SDL
SDL_CloseAudio();//Close SDL
SDL_Quit();
#endif
av_free(out_buffer);
avcodec_close(pCodecCtx);
avformat_close_input(&pFormatCtx);
return 0;
}