前言
首先再次说明下使用ffmpeg处理音视频流的步骤如下：1：从本地或者网络位置打开音视频流；
2：获取音视频流中的包，将其转换为帧；
3：如果帧不完整，转到2；
4：对帧做一些事情；
5：转到2；
我们所能做的就是在步骤4上进行一些处理。
本项目整体流程如下：
设置SDL的音频参数 —-> 打开声音设备，播放静音 —-> ffmpeg读取音频流中数据放入buffer —->SDL调用用户设置的函数（This function is called when the audio device needs more data.）来获取音频数据 —-> 播放音频注意：本项目中对音频的播放并没有做延时，但是声音是可以正常播放的。虽然程序将音频数据尽可能快的填充到声卡的缓冲区中，但是计算机的声音设备却是按照原来指定的采样率进行播放的。
源码
有了前篇文章ffmpeg+SDL的使用之使用SDL播放视频流的技术积累，现在需要学习的只是如何使用ffmpeg将音频数据存入buffer，使用回调函数进行播放。相对来讲还是比较简单的。头文件采用和前篇博文一样的头文件ffmpegsdl.h，工程配置方式也相同，可参考上篇博文，不再赘述。
audioPlay.cpp的源码如下：
#include <iostream>
#include "ffmpegsdl.h"
using namespace std;
#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit(4 byte) audio
static Uint8 *audio_chunk;
static int audio_len;
static Uint8 *audio_pos;
/**
* This function is called when the audio device needs more data.
*
* \param userdata An application-specific parameter saved in
* the SDL_AudioSpec structure
* \param stream A pointer to the audio data buffer.
* \param len The length of that buffer in bytes.
*
* Once the callback returns, the buffer will no longer be valid.
* Stereo samples are stored in a LRLRLR ordering.
*
* You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
* you like. Just open your audio device with a NULL callback.
*/
void audio_callback(void *udata, Uint8 *stream, int len)
{
//SDL 2.0
SDL_memset(stream, 0, len);
if (audio_len == 0)
{
return;
}
len = (len>audio_len ? audio_len : len); /* Mix as much data as possible */
/**
* This takes two audio buffers of the playing audio format and mixes
* them, performing addition, volume adjustment, and overflow clipping.
*/
SDL_MixAudio(stream, audio_pos, len, SDL_MIX_MAXVOLUME);
audio_pos += len;
audio_len -= len;
}
int main()
{
char pFile[] = "..\\test.mp4";
av_register_all();
AVFormatContext *pFormatCtx = avformat_alloc_context();
//Open
if (avformat_open_input(&pFormatCtx, pFile, 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, pFile, false);
AVCodec *pCodec;
// Find the first audio stream
int audio_stream_index = av_find_best_stream(pFormatCtx, AVMEDIA_TYPE_AUDIO, -1, -1, &pCodec, 0);
if (audio_stream_index == -1)
{
printf("Didn't find a audio stream.\n");
return -1;
}
if (pCodec == NULL)
{
printf("Codec not found.\n");
return -1;
}
AVCodecContext *pCodecCtx = avcodec_alloc_context3(pCodec);
//Fill the codec context based on the values from the supplied codec
avcodec_parameters_to_context(pCodecCtx, pFormatCtx->streams[audio_stream_index]->codecpar);
// Open codec
if (avcodec_open2(pCodecCtx, pCodec, NULL)<0)
{
printf("Could not open codec.\n");
return -1;
}
AVPacket* 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;
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);
uint8_t* out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE * 2);
AVFrame* pFrame = av_frame_alloc();
//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
SDL_AudioSpec wanted_spec;
wanted_spec.freq = pCodecCtx->sample_rate;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = pCodecCtx->channels;
wanted_spec.silence = 0;
wanted_spec.samples = pCodecCtx->frame_size;
wanted_spec.callback = audio_callback;
wanted_spec.userdata = pCodecCtx;
if (SDL_OpenAudio(&wanted_spec, NULL)<0)
{
printf("can't open audio.\n");
return -1;
}
//FIX:Some Codec's Context Information is missing
int64_t in_channel_layout = av_get_default_channel_layout(pCodecCtx->channels);
//Swr
SwrContext* 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);
int ret = -1;
int index = 0;
while (1)
{
ret = av_read_frame(pFormatCtx, packet);//一个包可能有多个帧
if (ret < 0)
{
break;
}
if (packet->stream_index == audio_stream_index)
{
//Supply raw packet data as input to a decoder.
ret = avcodec_send_packet(pCodecCtx, packet);
if (ret < 0)
{
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0)
{
//Return decoded output data from a decoder.
ret = avcodec_receive_frame(pCodecCtx, pFrame);
if (ret >= 0)
{
swr_convert(au_convert_ctx, &out_buffer, MAX_AUDIO_FRAME_SIZE, (const uint8_t **)pFrame->data, pFrame->nb_samples);
printf("index:%5d\t pts:%lld\t packet size:%d\n", index, packet->pts, packet->size);
index++;
}
else if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
{
break;
}
else
{
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
}
while (audio_len > 0)//Wait until finish
{
SDL_Delay(1);
}
//Set audio buffer (PCM data)
audio_chunk = (Uint8 *)out_buffer;
//Audio buffer length
audio_len = out_buffer_size;
audio_pos = audio_chunk;
}
av_packet_unref(packet);
}
end:
// Free the packet that was allocated by av_read_frame
av_packet_free(&packet);
swr_free(&au_convert_ctx);
SDL_CloseAudio();//Close SDL
SDL_Quit();
av_free(out_buffer);
avcodec_close(pCodecCtx);
avformat_close_input(&pFormatCtx);
return 0;
}
工程下载
github: https://github.com/zhuyinglong/ffmpeg-SDL
后续
虽然我尽力让自己正确，但避免不了有些理解不到位的误导大家，有些不清楚的地方，推荐大家还是参考官方原版一手的资料。参考资料如下：
http://dranger.com/ffmpeg/tutorial03.html
http://ffmpeg.org/doxygen/trunk/filtering_video_8c-example.html