why should we debug your code?Verzonden vanaf mijn Samsung Galaxy-smartphone.
-------- Oorspronkelijk bericht --------Van: 강구철 <[email protected]> Datum:
23-11-18 07:38 (GMT+01:00) Aan: "'This list is about using libavcodec,
libavformat, libavutil, libavdevice and libavfilter.'" <[email protected]>
Onderwerp: [Libav-user] transcoder class instance memory leak. hear is
transcoder class src. use it as a instance and memory leak after delete
instance.every resource release and close. what's problem? main transact func
is "tcode(opus pkt in, aac pkt out)"class trans{ public: ///DECLARE
AVCodecContext *oc; AVCodecContext *opus_context; AVCodecContext
*aac_context; AVCodecContext *aac_context2; AVCodecContext
*input_codec_context; AVFormatContext *output_format_context; const
AVCodec *opus_codec; const AVCodec *aac_codec; const AVCodec
*aac_codec2; const AVCodec *h264_codec; float AAC_ENCTIME; int
AAC_SAMPRATE; int OPUS_SAMPRATE; int OPUSsize; int
OPUScnt ; int AACsize ; int
AACcnt ; AVFrame *decoded_frame;///opus decode by ffmpeg
AVFrame *decoded_frame2;///opus decode by ffmpeg AVFrame
*framebuf; AVPacket *pkt1; AVPacket *pkt2; int
data_present; CUnitInspection *gen1; CUnitInspection *gen2;
AVAudioFifo *fifo; AVAudioFifo *fifo2; SwrContext *resample_context;
RTPPaserUtilityLibrary::ConvertStreamAVGranules* _instance; int len;
string outfile; int pcmsize; uint8_t inbuf[AUDIO_INBUF_SIZE +
AV_INPUT_BUFFER_PADDING_SIZE]; char out_buf [2000]; //from file int
out_size; uint8_t *data; size_t data_size; int ix1; int
channel; int ret; int cnt[100]; int init(int a, int b, int c);
trans(); ~trans(); int push_q(AVAudioFifo *fifo,
uint8_t **converted_input_samples, const int
frame_size); int pop_q(AVAudioFifo *_queue, AVFrame* mframe); int
tcode(string *src, string* dst); int tcode(AVPacket* src, AVPacket*
dst);};int RTP_REASSEMBLE::trans::init(int a, int b, int c){
opus_context->channels = 1; opus_context->channel_layout =
AV_CH_LAYOUT_MONO; opus_context->sample_rate = OPUS_SAMPRATE;
opus_context->sample_fmt = AV_SAMPLE_FMT_FLTP;
opus_context->bit_rate = OPUS_SAMPRATE; aac_context->channels = 1;
aac_context->channel_layout = AV_CH_LAYOUT_MONO;
aac_context->sample_rate = AAC_SAMPRATE; aac_context->sample_fmt
= AV_SAMPLE_FMT_FLTP; aac_context->bit_rate = AAC_SAMPRATE;
fifo = av_audio_fifo_alloc(aac_context->sample_fmt, aac_context->channels,
2048); aac_context2->channels = 1; aac_context2->channel_layout =
AV_CH_LAYOUT_MONO; aac_context2->sample_rate = AAC_SAMPRATE;
aac_context2->sample_fmt = AV_SAMPLE_FMT_FLTP;
aac_context2->bit_rate = AAC_SAMPRATE; return
0;};RTP_REASSEMBLE::trans::trans(){ oc = NULL;
opus_context = NULL; aac_context =
NULL; input_codec_context = NULL; output_format_context
= NULL; OPUSsize = 0; OPUScnt
= 0; AACsize = 0; h264_codec
= NULL; AACcnt = 0; decoded_frame
= NULL;///opus decode by ffmpeg decoded_frame2 =
NULL;///opus decode by ffmpeg fifo = NULL;
//fifo2 = NULL; resample_context =
NULL; AAC_SAMPRATE=48000; OPUS_SAMPRATE=48000; ix1=0;
ret = AVERROR_EXIT; avcodec_register_all(); h264_codec =
avcodec_find_decoder( AV_CODEC_ID_H264 ); opus_codec =
avcodec_find_decoder( AV_CODEC_ID_OPUS ); aac_codec =
avcodec_find_encoder( AV_CODEC_ID_AAC ); aac_codec2 =
avcodec_find_decoder( AV_CODEC_ID_AAC );// ENCODER //aac_codec =
avcodec_find_encoder( V_CODEC_ID_AAC_LATM ); framebuf =
av_frame_alloc(); if(framebuf==NULL) cout << "frame alloc fail."<<endl;
_instance = new RTPPaserUtilityLibrary::ConvertStreamAVGranules();
pkt1 = av_packet_alloc(); pkt2 = av_packet_alloc(); if
(!h264_codec) { cout << "H264 Codec not found"<<endl;
return; } else cout << "H264 codec found."<<endl;
if(framebuf==NULL) cout << "frame alloc fail."<<endl;/* gen1 = new
CUnitInspection(); gen2 = new CUnitInspection(); gen1-> ifile =
"opus.rtp"; gen1->_begin(); gen2-> ifile = "264.rtp";
gen2->_begin();*/ if (!opus_codec) { cout << "opus Codec not
found"<<endl; return; } else cout << "OPUS codec
found."<<endl; opus_context = avcodec_alloc_context3( opus_codec );
//_instance->init_fifo(&fifo, opus_context); if (!opus_context) {
cout << "Could not allocate audio opus_codec context OPUS"<<endl;
return; } outfile = "raw.mp4"; if (!aac_codec) {
cout << "AAC Codec not found"<<endl; return; }
else cout << "AAC codec found."<<endl; aac_context =
avcodec_alloc_context3(aac_codec); //_instance->init_fifo(&fifo2,
aac_context); if (!aac_context) { cout << "Could not allocate
audio codec context AAC"<<endl; return; } aac_context2
= avcodec_alloc_context3(aac_codec2); init(1,2,3); if
(avcodec_open2(opus_context, opus_codec, NULL) < 0) { cout << "Could
not open codec opus"<<endl; return; } if
(avcodec_open2(aac_context, aac_codec, NULL) < 0) { cout << "Could
not open codec aac"<<endl; return; } if
(avcodec_open2(aac_context2, aac_codec2, NULL) < 0) { cout << "Could
not open codec aac2"<<endl; return; } if
(!decoded_frame) { cout << "Got a decoded_frame"<<endl;
if (!(decoded_frame = av_frame_alloc())) { cout << "Could
not allocate audio frame"<<endl; return; }
}else{ cout << "nodata decoded_frame"<<endl; }
if (!decoded_frame2) { cout << "Got a
decoded_frame2"<<endl; if (!(decoded_frame2 = av_frame_alloc())) {
cout << "Could not allocate audio frame"<<endl;
return; } }else{ cout << "nodata
decoded_frame2"<<endl;
}}RTP_REASSEMBLE::trans::~trans(){OsSysLog::add(FAC_PROCESS, PRI_INFO,
"%s:%d:FREE FIFO************************* ",__FILE__,__LINE__); // delete
gen1; // delete gen2; av_free_packet(pkt1);
av_free_packet(pkt2); av_frame_free(&decoded_frame);
av_frame_free(&decoded_frame2); av_audio_fifo_free(fifo);
//av_audio_fifo_free(fifo2); delete _instance;}/** Add converted input
audio samples to the FIFO buffer for later processing. */int
RTP_REASSEMBLE::trans::push_q(AVAudioFifo *fifo,
uint8_t **converted_input_samples, const int
frame_size){ int error; /** * Make the FIFO as large as it needs to
be to hold both, * the old and the new samples. */
//OsSysLog::add(FAC_PROCESS, PRI_INFO, "%s:%d:*************************PUSH
called fifo:0x%x, data:0x%x, frame_size:%d",__FILE__,__LINE__,(int64_t)fifo,
(void*)**converted_input_samples, frame_size); if ((error =
av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
return error; } /** Store the new samples in the FIFO buffer. */ if
(av_audio_fifo_write(fifo, (void **)converted_input_samples, frame_size) <
frame_size) { return AVERROR_EXIT; }
//OsSysLog::add(FAC_PROCESS, PRI_INFO, "%s:%d:*************************OK
PUSH:%d tot:%d", __FILE__, __LINE__, frame_size, av_audio_fifo_size(fifo));
return 1;}int RTP_REASSEMBLE::trans::pop_q(AVAudioFifo *_fifo, AVFrame*
mframe){ int ret; if( av_audio_fifo_size (_fifo) >= 1024) {
void* ptr; ptr = * mframe->data; ret =
av_audio_fifo_read(_fifo, (void**)mframe->data, 1024);//@#@#
mframe->nb_samples = ret; return ret; } else {
//OsSysLog::add(FAC_PROCESS, PRI_INFO, "%s:%d:*Not enough data popup ret:%d
tot:%d",__FILE__,__LINE__, ret, av_audio_fifo_size(_fifo)); return 0;
}}int RTP_REASSEMBLE::trans::tcode(AVPacket* src, AVPacket* dst){//return
-1;//KKC 8888 AACContext *ac = (AACContext*)aac_context->priv_data; /*
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int
*got_frame_ptr, GetBitContext *gb) { AACContext *ac =
avctx->priv_data; const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
ChannelElement *che; int err, i; int samples =
m4ac->frame_length_short ? 960 : 1024; int chan_config =
m4ac->chan_config; int aot = m4ac->object_type; if (aot ==
AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) samples >>= 1; }*/
MPEG4AudioConfig *m4ac = &(ac->oc[0].m4ac); m4ac->frame_length_short =
1;//1:960, 0:1024 int ret; FILE *f, *outfile; int data_size; data_size
= av_get_bytes_per_sample(opus_context->sample_fmt); if (src->size){
data_present = 0; //decoded_frame->linesize[0]=1024*4;
//decoded_frame->linesize[1]=1024*4;
//opus_context->frame_size=960;//TESTTEST //decoded_frame->data[0] =
(uint8_t*)av_malloc(4*1024);//TESTTEST if(decoded_frame2->data[0]!=NULL)
av_free(decoded_frame2->data[0]); decoded_frame2->data[0] =
(uint8_t*)av_malloc(4*1024);//@#@# ret = avcodec_decode_audio4(opus_context,
decoded_frame, &data_present, src); if(ret<0) {
//OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d DECODE Error ======%d", __FILE__,
__LINE__,ret); return ret; }#ifdef _NOUSE else {
if(data_present){ if(channel%2)//7777
outfile = fopen("/temp/abc1.wav", "a"); else
outfile = fopen("/temp/abc2.wav", "a"); if (!outfile) {
OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d abc.wav open Error ======",
__FILE__, __LINE__); //return 0; }
//av_freep(&decoded_frame);//TESTTEST unsigned char
cbuf[4]={0,0,0,0};//GEN WAVE------------------; unsigned char *pbuf;
cbuf[0]=0xf2; cbuf[1]=0xdb; cbuf[2]=0x0;
cbuf[3]=0x3f; //decoded_frame->data[0] =
(uint8_t*)av_malloc(4*1024);//BLANK 1024 //if(1)
decoded_frame->nb_samples = 960;//good for mp4 //else
decoded_frame->nb_samples = 1024;//good for wav space
pbuf = decoded_frame->data[0];#ifdef _NOUSE for(int i=960; i<1024 ;
i++){// OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d i:%d ======",
__FILE__, __LINE__, i*4+3); pbuf[i*4+0] = cbuf[0]; //float
pbuf[i*4+1] = cbuf[1]; //float pbuf[i*4+2] =
cbuf[2]; //float pbuf[i*4+3] = cbuf[3]; //float
cbuf[2] += 2; //mod wav to 0~96 if(cbuf[2] >= 96)
cbuf[2]=0; //mod reset to 0 }//END OF
GENWAVE---------------------------------------------------------#endif
fwrite(decoded_frame->data[0] , decoded_frame->nb_samples*data_size, 1,
outfile);//opus_context->channels* fclose(outfile);
//OsSysLog::add(FAC_PROCESS,PRI_INFO,"LN:%d Ddata_present=%d OPUS DECODE
Success === duration:%d ret(size):%d linesize:%d nbsampels:%d", __LINE__,
data_present, decoded_frame->pkt_duration,
ret,decoded_frame->linesize,decoded_frame->nb_samples); }//data present
growes up and cut off it. }#endif //#ifdef _VIDEODUMP// test
file write//OsSysLog::add(FAC_PROCESS, PRI_INFO, "BAAAAAAAAAAAAAAAAAAAAA
ch=%d,samp=%d, fmt=%d, data_size=%d ch=%d chLayout=%ld frame_size=%d\n", \
opus_context->channels, decoded_frame->nb_samples, opus_context->sample_fmt,
data_size, opus_context->channels,
opus_context->channel_layout,opus_context->frame_size); OPUSsize +=
decoded_frame->nb_samples * opus_context->channels *
data_size;//OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d RAW PCMsize= %d, opusCNT=
%d decodeRAW:%d", __FILE__, __LINE__, OPUSsize, OPUScnt++,
decoded_frame->nb_samples);//#endif av_init_packet(dst); dst->data =
NULL;//IMPORTANT for dynamic reallocate surface dst->size = 0;ret = -1;
///////////////////////////////////////////////////////////////////////////////
//int push_q(AVAudioFifo *fifo, uint8_t **converted_input_samples,
const int frame_size); //int pop_q(AVAudioFifo *fifo, AVFrame* mframe);
/////////////////////////////////////////////////////////////////////////////
uint8_t* ptr; //ptr = push_q(fifo, decoded_frame->data,
decoded_frame->nb_samples);//PUSH decoded_frame2->nb_samples = 1024;
ret = pop_q (fifo, decoded_frame2); //POP
//decoded_frame->nb_samples = 1024; //aac_context->frame_size = 960;
aac_context->frame_size = 1024;
//OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d AAC ENCODE try decodedframeSize:%d
aacctxFramesieze:%d======frameLnShort:%d privdataSIZE:0x%x", __FILE__,
__LINE__, decoded_frame2->nb_samples, aac_context->frame_size,
m4ac->frame_length_short,aac_context->priv_data); if(ret>0){// pop ok (que
hav sample over 1024) ret = avcodec_encode_audio2(aac_context, dst,
decoded_frame2, &data_present);//@#@#AAC ENCODING#ifdef
_AAC_ENCODE_RESULT_PRINT if(ret != 0) {
OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d AAC ENCODE Error OK ret:%d ======",
__FILE__, __LINE__, ret); }else{
OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d AAC ENCODE OK size is %d AACcnt:%d
AACsize:%d aacctxFramesize:%d======", __FILE__, __LINE__,
dst->size, AACcnt, AACsize,aac_context->frame_size); AACsize +=
(int)(dst->size); if(pkt2->size != 0) AACcnt++;
}#endif }}#ifdef _DECODE_VERIFY_AAC ret =
avcodec_decode_audio4(aac_context2, decoded_frame, &data_present, dst);
//if(ret<1) OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d AAC REDECODE Error
======%d", __FILE__, __LINE__,ret);#endif#ifdef _DUMP_REDECODE if(ret<0)
{ OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d AAC REDECODE Error
======%d", __FILE__, __LINE__,ret); //return ret; }else{
if(data_present){ //OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d AAC
REDECODE ====%d frsize:%d", __FILE__, __LINE__,ret,
decoded_frame->nb_samples); if(channel%2)//7777
outfile = fopen("/temp/aac1.wav", "a+"); else outfile
= fopen("/temp/aac2.wav", "a+"); if (!outfile) {
OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d aac.wav open Error ======",
__FILE__, __LINE__); return 0; }
fwrite(decoded_frame->data[0] , /*decoded_frame->nb_samples *
opus_context->channels */1024 * data_size, 1,
outfile);//opus_context->channels* fclose(outfile);
OsSysLog::add(FAC_PROCESS,PRI_INFO,"LN:%d Ddata_present=%d DECODE Success ===
duration:%d ret(size):%d linesize:%d nbsampels:%d", __LINE__, data_present,
decoded_frame->pkt_duration,
ret,decoded_frame->linesize,decoded_frame->nb_samples); } else{
OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d DECODE no presents data!!
======", __FILE__, __LINE__); return ret; } }#endif
return 0;}int RTP_REASSEMBLE::trans::tcode(string *src, string* dst){
if(pkt1 == NULL) { OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d
Error pkt1 NULL====data_size[%d]======", __FILE__, __LINE__, src->length());
init(1,2,3); } // COPY NEED , dont use STRING
pkt1->data = (uint8_t*)src->c_str(); pkt1->size = src->length();//solve
source packet of opus sound encoded data //memcpy(pkt1->data, src->c_str(),
src->length()); tcode(pkt1,pkt2); if(pkt2 == NULL) {
OsSysLog::add(FAC_PROCESS,PRI_INFO,"%s:%d Error pkt2
NUL======data22_size[%d]=====", __FILE__, __LINE__, dst->length());
init(1,2,3); } dst->assign((uint8_t&)pkt2->data, pkt2->size);//build
dst if(pkt2->size==0){ OsSysLog::add(FAC_PROCESS,PRI_INFO, "%s:%d
Error====data2_size zero[%d:%d]=======", __FILE__, __LINE__, src->length(),
dst->length()); } return 1;// protection code
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