我关注了iFrameExtractor在我的 swift 项目中成功流式传输 rtsp。在这个项目中，它还具有录音功能。它基本上使用avformat_write_header, av_interleaved_write_frame和 av_write_trailer将 rtsp 源保存到 mp4 文件中。
当我在我的设备中使用这个项目时，rtsp 流工作正常，但录制功能总是会生成一个没有图像和声音的空白 mp4 文件。
谁能告诉我我错过了哪一步？
我将 iPhone5 与 iOS 9.1 和 XCode 7.1.1 一起使用。
ffmpeg 是 2.8.3 版本，编译指令为 CompilationGuide – FFmpeg
以下是本项目中的示例代码
生成每一帧的函数:
-(BOOL)stepFrame {
// AVPacket packet;
int frameFinished=0;
static bool bFirstIFrame=false;
static int64_t vPTS=0, vDTS=0, vAudioPTS=0, vAudioDTS=0;
while(!frameFinished && av_read_frame(pFormatCtx, &packet)>=0) {
// Is this a packet from the video stream?
if(packet.stream_index==videoStream) {
// 20130525 albert.liao modified start
// Initialize a new format context for writing file
if(veVideoRecordState!=eH264RecIdle)
{
switch(veVideoRecordState)
{
case eH264RecInit:
{
if ( !pFormatCtx_Record )
{
int bFlag = 0;
//NSString *videoPath = [NSHomeDirectory() stringByAppendingPathComponent:@"Documents/test.mp4"];
NSString *videoPath = @"/Users/liaokuohsun/iFrameTest.mp4";
const char *file = [videoPath UTF8String];
pFormatCtx_Record = avformat_alloc_context();
bFlag = h264_file_create(file, pFormatCtx_Record, pCodecCtx, pAudioCodecCtx,/*fps*/0.0, packet.data, packet.size );
if(bFlag==true)
{
veVideoRecordState = eH264RecActive;
fprintf(stderr, "h264_file_create success\n");
}
else
{
veVideoRecordState = eH264RecIdle;
fprintf(stderr, "h264_file_create error\n");
}
}
}
//break;
case eH264RecActive:
{
if((bFirstIFrame==false) &&(packet.flags&AV_PKT_FLAG_KEY)==AV_PKT_FLAG_KEY)
{
bFirstIFrame=true;
vPTS = packet.pts ;
vDTS = packet.dts ;
#if 0
NSRunLoop *pRunLoop = [NSRunLoop currentRunLoop];
[pRunLoop addTimer:RecordingTimer forMode:NSDefaultRunLoopMode];
#else
[NSTimer scheduledTimerWithTimeInterval:5.0//2.0
target:self
selector:@selector(StopRecording:)
userInfo:nil
repeats:NO];
#endif
}
// Record audio when 1st i-Frame is obtained
if(bFirstIFrame==true)
{
if ( pFormatCtx_Record )
{
#if PTS_DTS_IS_CORRECT==1
packet.pts = packet.pts - vPTS;
packet.dts = packet.dts - vDTS;
#endif
h264_file_write_frame( pFormatCtx_Record, packet.stream_index, packet.data, packet.size, packet.dts, packet.pts);
}
else
{
NSLog(@"pFormatCtx_Record no exist");
}
}
}
break;
case eH264RecClose:
{
if ( pFormatCtx_Record )
{
h264_file_close(pFormatCtx_Record);
#if 0
// 20130607 Test
dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^(void)
{
ALAssetsLibrary *library = [[ALAssetsLibrary alloc]init];
NSString *filePathString = [NSHomeDirectory() stringByAppendingPathComponent:@"Documents/test.mp4"];
NSURL *filePathURL = [NSURL fileURLWithPath:filePathString isDirectory:NO];
if(1)// ([library videoAtPathIsCompatibleWithSavedPhotosAlbum:filePathURL])
{
[library writeVideoAtPathToSavedPhotosAlbum:filePathURL completionBlock:^(NSURL *assetURL, NSError *error){
if (error) {
// TODO: error handling
NSLog(@"writeVideoAtPathToSavedPhotosAlbum error");
} else {
// TODO: success handling
NSLog(@"writeVideoAtPathToSavedPhotosAlbum success");
}
}];
}
[library release];
});
#endif
vPTS = 0;
vDTS = 0;
vAudioPTS = 0;
vAudioDTS = 0;
pFormatCtx_Record = NULL;
NSLog(@"h264_file_close() is finished");
}
else
{
NSLog(@"fc no exist");
}
bFirstIFrame = false;
veVideoRecordState = eH264RecIdle;
}
break;
default:
if ( pFormatCtx_Record )
{
h264_file_close(pFormatCtx_Record);
pFormatCtx_Record = NULL;
}
NSLog(@"[ERROR] unexpected veVideoRecordState!!");
veVideoRecordState = eH264RecIdle;
break;
}
}
// Decode video frame
avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
}
else if(packet.stream_index==audioStream)
{
// 20131024 albert.liao modfied start
static int vPktCount=0;
BOOL bIsAACADTS = FALSE;
int ret = 0;
if(aPlayer.vAACType == eAAC_UNDEFINED)
{
tAACADTSHeaderInfo vxAACADTSHeaderInfo = {0};
bIsAACADTS = [AudioUtilities parseAACADTSHeader:(uint8_t *)packet.data ToHeader:&vxAACADTSHeaderInfo];
}
@synchronized(aPlayer)
{
if(aPlayer==nil)
{
aPlayer = [[AudioPlayer alloc]initAudio:nil withCodecCtx:(AVCodecContext *) pAudioCodecCtx];
NSLog(@"aPlayer initAudio");
if(bIsAACADTS)
{
aPlayer.vAACType = eAAC_ADTS;
//NSLog(@"is ADTS AAC");
}
}
else
{
if(vPktCount<5) // The voice is listened once image is rendered
{
vPktCount++;
}
else
{
if([aPlayer getStatus]!=eAudioRunning)
{
dispatch_async(dispatch_get_main_queue(), ^(void) {
@synchronized(aPlayer)
{
NSLog(@"aPlayer start play");
[aPlayer Play];
}
});
}
}
}
};
@synchronized(aPlayer)
{
int ret = 0;
ret = [aPlayer putAVPacket:&packet];
if(ret <= 0)
NSLog(@"Put Audio Packet Error!!");
}
// 20131024 albert.liao modfied end
if(bFirstIFrame==true)
{
switch(veVideoRecordState)
{
case eH264RecActive:
{
if ( pFormatCtx_Record )
{
h264_file_write_audio_frame(pFormatCtx_Record, pAudioCodecCtx, packet.stream_index, packet.data, packet.size, packet.dts, packet.pts);
}
else
{
NSLog(@"pFormatCtx_Record no exist");
}
}
}
}
}
else
{
//fprintf(stderr, "packet len=%d, Byte=%02X%02X%02X%02X%02X\n",\
packet.size, packet.data[0],packet.data[1],packet.data[2],packet.data[3], packet.data[4]);
}
// 20130525 albert.liao modified end
}
return frameFinished!=0;
}
avformat_write_header:
int h264_file_create(const char *pFilePath, AVFormatContext *fc, AVCodecContext *pCodecCtx,AVCodecContext *pAudioCodecCtx, double fps, void *p, int len )
{
int vRet=0;
AVOutputFormat *of=NULL;
AVStream *pst=NULL;
AVCodecContext *pcc=NULL, *pAudioOutputCodecContext=NULL;
avcodec_register_all();
av_register_all();
av_log_set_level(AV_LOG_VERBOSE);
if(!pFilePath)
{
fprintf(stderr, "FilePath no exist");
return -1;
}
if(!fc)
{
fprintf(stderr, "AVFormatContext no exist");
return -1;
}
fprintf(stderr, "file=%s\n",pFilePath);
// Create container
of = av_guess_format( 0, pFilePath, 0 );
fc->oformat = of;
strcpy( fc->filename, pFilePath );
// Add video stream
pst = avformat_new_stream( fc, 0 );
vVideoStreamIdx = pst->index;
fprintf(stderr,"Video Stream:%d",vVideoStreamIdx);
pcc = pst->codec;
avcodec_get_context_defaults3( pcc, AVMEDIA_TYPE_VIDEO );
// Save the stream as origin setting without convert
pcc->codec_type = pCodecCtx->codec_type;
pcc->codec_id = pCodecCtx->codec_id;
pcc->bit_rate = pCodecCtx->bit_rate;
pcc->width = pCodecCtx->width;
pcc->height = pCodecCtx->height;
if(fps==0)
{
double fps=0.0;
AVRational pTimeBase;
pTimeBase.num = pCodecCtx->time_base.num;
pTimeBase.den = pCodecCtx->time_base.den;
fps = 1.0/ av_q2d(pCodecCtx->time_base)/ FFMAX(pCodecCtx->ticks_per_frame, 1);
fprintf(stderr,"fps_method(tbc): 1/av_q2d()=%g",fps);
pcc->time_base.num = 1;
pcc->time_base.den = fps;
}
else
{
pcc->time_base.num = 1;
pcc->time_base.den = fps;
}
// reference ffmpeg\libavformat\utils.c
// For SPS and PPS in avcC container
pcc->extradata = malloc(sizeof(uint8_t)*pCodecCtx->extradata_size);
memcpy(pcc->extradata, pCodecCtx->extradata, pCodecCtx->extradata_size);
pcc->extradata_size = pCodecCtx->extradata_size;
// For Audio stream
if(pAudioCodecCtx)
{
AVCodec *pAudioCodec=NULL;
AVStream *pst2=NULL;
pAudioCodec = avcodec_find_encoder(AV_CODEC_ID_AAC);
// Add audio stream
pst2 = avformat_new_stream( fc, pAudioCodec );
vAudioStreamIdx = pst2->index;
pAudioOutputCodecContext = pst2->codec;
avcodec_get_context_defaults3( pAudioOutputCodecContext, pAudioCodec );
fprintf(stderr,"Audio Stream:%d",vAudioStreamIdx);
fprintf(stderr,"pAudioCodecCtx->bits_per_coded_sample=%d",pAudioCodecCtx->bits_per_coded_sample);
pAudioOutputCodecContext->codec_type = AVMEDIA_TYPE_AUDIO;
pAudioOutputCodecContext->codec_id = AV_CODEC_ID_AAC;
// Copy the codec attributes
pAudioOutputCodecContext->channels = pAudioCodecCtx->channels;
pAudioOutputCodecContext->channel_layout = pAudioCodecCtx->channel_layout;
pAudioOutputCodecContext->sample_rate = pAudioCodecCtx->sample_rate;
pAudioOutputCodecContext->bit_rate = 12000;//pAudioCodecCtx->sample_rate * pAudioCodecCtx->bits_per_coded_sample;
pAudioOutputCodecContext->bits_per_coded_sample = pAudioCodecCtx->bits_per_coded_sample;
pAudioOutputCodecContext->profile = pAudioCodecCtx->profile;
//FF_PROFILE_AAC_LOW;
// pAudioCodecCtx->bit_rate;
// AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P
//pAudioOutputCodecContext->sample_fmt = AV_SAMPLE_FMT_FLTP;//pAudioCodecCtx->sample_fmt;
pAudioOutputCodecContext->sample_fmt = pAudioCodecCtx->sample_fmt;
//pAudioOutputCodecContext->sample_fmt = AV_SAMPLE_FMT_U8;
pAudioOutputCodecContext->sample_aspect_ratio = pAudioCodecCtx->sample_aspect_ratio;
pAudioOutputCodecContext->time_base.num = pAudioCodecCtx->time_base.num;
pAudioOutputCodecContext->time_base.den = pAudioCodecCtx->time_base.den;
pAudioOutputCodecContext->ticks_per_frame = pAudioCodecCtx->ticks_per_frame;
pAudioOutputCodecContext->frame_size = 1024;
fprintf(stderr,"profile:%d, sample_rate:%d, channles:%d", pAudioOutputCodecContext->profile, pAudioOutputCodecContext->sample_rate, pAudioOutputCodecContext->channels);
AVDictionary *opts = NULL;
av_dict_set(&opts, "strict", "experimental", 0);
if (avcodec_open2(pAudioOutputCodecContext, pAudioCodec, &opts) < 0) {
fprintf(stderr, "\ncould not open codec\n");
}
av_dict_free(&opts);
#if 0
// For Audio, this part is no need
if(pAudioCodecCtx->extradata_size!=0)
{
NSLog(@"extradata_size !=0");
pAudioOutputCodecContext->extradata = malloc(sizeof(uint8_t)*pAudioCodecCtx->extradata_size);
memcpy(pAudioOutputCodecContext->extradata, pAudioCodecCtx->extradata, pAudioCodecCtx->extradata_size);
pAudioOutputCodecContext->extradata_size = pAudioCodecCtx->extradata_size;
}
else
{
// For WMA test only
pAudioOutputCodecContext->extradata_size = 0;
NSLog(@"extradata_size ==0");
}
#endif
}
if(fc->oformat->flags & AVFMT_GLOBALHEADER)
{
pcc->flags |= CODEC_FLAG_GLOBAL_HEADER;
pAudioOutputCodecContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
if ( !( fc->oformat->flags & AVFMT_NOFILE ) )
{
vRet = avio_open( &fc->pb, fc->filename, AVIO_FLAG_WRITE );
if(vRet!=0)
{
fprintf(stderr,"avio_open(%s) error", fc->filename);
}
}
// dump format in console
av_dump_format(fc, 0, pFilePath, 1);
vRet = avformat_write_header( fc, NULL );
if(vRet==0)
return 1;
else
return 0;
}
av_interleaved_write_frame:
void h264_file_write_frame(AVFormatContext *fc, int vStreamIdx, const void* p, int len, int64_t dts, int64_t pts )
{
AVStream *pst = NULL;
AVPacket pkt;
if ( 0 > vVideoStreamIdx )
return;
// may be audio or video
pst = fc->streams[ vStreamIdx ];
// Init packet
av_init_packet( &pkt );
if(vStreamIdx ==vVideoStreamIdx)
{
pkt.flags |= ( 0 >= getVopType( p, len ) ) ? AV_PKT_FLAG_KEY : 0;
//pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = pst->index;
pkt.data = (uint8_t*)p;
pkt.size = len;
pkt.dts = AV_NOPTS_VALUE;
pkt.pts = AV_NOPTS_VALUE;
// TODO: mark or unmark the log
//fprintf(stderr, "dts=%lld, pts=%lld\n",dts,pts);
// av_write_frame( fc, &pkt );
}
av_interleaved_write_frame( fc, &pkt );
}
av_write_trailer:
void h264_file_close(AVFormatContext *fc)
{
if ( !fc )
return;
av_write_trailer( fc );
if ( fc->oformat && !( fc->oformat->flags & AVFMT_NOFILE ) && fc->pb )
avio_close( fc->pb );
av_free( fc );
}
谢谢。
最佳答案
看起来您对输入和输出使用相同的 AVFormatContext ？
在行
pst = fc->streams[ vStreamIdx ];
您正在从与您的输入(RTSP 流)连接的 AVFormatContext 分配 AVStream*。但是稍后您会尝试将数据包写回相同的上下文
av_interleaved_write_frame( fc, &pkt ); .我有点将上下文视为一个文件，它帮助我更好地导航这类事情。我做了一些与你正在做的事情相同的事情(虽然不是 iOS)，我为每个输入(RTSP 流)和输出(mp4 文件)使用单独的 AVFormatContext。如果我是正确的，我认为您只需要正确初始化 AVFormatContext 即可。以下代码(没有错误检查所有内容)是我为获取
AVFormatContext * output_format_context = NULL 所做的操作和 AVFormatContext * input_format_context我已经与 RTSP 流相关联并从一个写入另一个。这是在我获取数据包等之后，在您的情况下，它看起来像您正在填充(我只是从 av_read_frame 获取数据包并重新打包它。这是可能在您的 write frame 函数中的代码(但它也包括标题的写入)。
AVFormatContext * output_format_context;
AVStream * in_stream_2;
AVStream * out_stream_2;
// Allocate the context with the output file
avformat_alloc_output_context2(&output_format_context, NULL, NULL, out_filename.c_str());
// Point to AVOutputFormat * output_format for manipulation
output_format = output_format_context->oformat;
// Loop through all streams
for (i = 0; i < input_format_context->nb_streams; i++) {
// Create a pointer to the input stream that was allocated earlier in the code
AVStream *in_stream = input_format_context->streams[i];
// Create a pointer to a new stream that will be part of the output
AVStream *out_stream = avformat_new_stream(output_format_context, in_stream->codec->codec);
// Set time_base of the new output stream to equal the input stream one since I'm not changing anything (can avoid but get a deprecation warning)
out_stream->time_base = in_stream->time_base;
// This is the non-deprecated way of copying all the parameters from the input stream into the output stream since everything stays the same
avcodec_parameters_from_context(out_stream->codecpar, in_stream->codec);
// I don't remember what this is for :)
out_stream->codec->codec_tag = 0;
// This just sets a flag from the format context to the stream relating to the header
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
// Check NOFILE flag and open the output file context (previously the output file was associated with the format context, now it is actually opened.
if (!(output_format->flags & AVFMT_NOFILE))
avio_open(&output_format_context->pb, out_filename.c_str(), AVIO_FLAG_WRITE);
// Write the header (not sure if this is always needed but h264 I believe it is.
avformat_write_header(output_format_context,NULL);
// Re-getting the appropriate stream that was populated above (this should allow for both audio/video)
in_stream_2 = input_format_context->streams[packet.stream_index];
out_stream_2 = output_format_context->streams[packet.stream_index];
// Rescaling pts and dts, duration and pos - you would do as you need in your code.
packet.pts = av_rescale_q_rnd(packet.pts, in_stream_2->time_base, out_stream_2->time_base, (AVRounding) (AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
packet.dts = av_rescale_q_rnd(packet.dts, in_stream_2->time_base, out_stream_2->time_base, (AVRounding) (AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
packet.duration = av_rescale_q(packet.duration, in_stream_2->time_base, out_stream_2->time_base);
packet.pos = -1;
// The first packet of my stream always gives me negative dts/pts so this just protects that first one for my purposes. You probably don't need.
if (packet.dts < 0) packet.dts = 0;
if (packet.pts < 0) packet.pts = 0;
// Finally write the frame
av_interleaved_write_frame(output_format_context, &packet);
// ....
// Write header, close/cleanup... etc
// ....
这段代码相当简单，不包括设置(听起来你无论如何都做得正确)。我还想象可以根据您的目的清理和调整此代码，但这对我来说可以将 RTSP 流重新写入文件(在我的情况下很多文件但未显示代码)。
该代码是 C 代码，因此您可能需要进行一些小的调整以使其与 Swift 兼容(可能对于某些库函数调用)。我认为总的来说它应该是兼容的。
希望这有助于为您指明正确的方向。这要归功于几个示例代码源(我不记得在哪里)，以及来自库本身的警告提示。
关于objective-c - 在 iOS 中使用 ffmpeg 录制 rtsp 流，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/34193474/