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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
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Sur d’autres sites (9870)
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Wav File encoded with FFMPEG has issues with codecs while playing using VLC Player
24 mai 2022, par user924702I want to convert raw PCM data(Taken from Android Phone mic) into a libGSM Wave file. After encoding into file, VLC player shows right codec information and duration but unable to play contents. Please help me to find what I am doing wrong.



Below is my code for encoding and header writing :



void EncodeTest(uint8_t *audioData, size_t audioSize)
{
 AVCodecContext *audioCodec;
 AVCodec *codec;
 uint8_t *buf; int bufSize, frameBytes;
 __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"Lets encode :%u with size %d\n",(int)audioData, (int)audioSize);
 //Set up audio encoder
 codec = avcodec_find_encoder(CODEC_ID_GSM);
 if (codec == NULL){
 __android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to find encoder(CODEC_ID_GSM)");
 codec = avcodec_find_encoder(CODEC_ID_GSM);
 if (codec == NULL){
 __android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to find encoder(CODEC_ID_GSM)");
 return;
 }
 }
 audioCodec = avcodec_alloc_context();
 audioCodec->channels = 1;
 audioCodec->sample_rate = 8000;
 audioCodec->sample_fmt = SAMPLE_FMT_S16;
 audioCodec->bit_rate = 13200;
 audioCodec->priv_data = gsm_create();

 switch(audioCodec->codec_id) {
 case CODEC_ID_GSM:
 audioCodec->frame_size = GSM_FRAME_SIZE;
 audioCodec->block_align = GSM_BLOCK_SIZE;
 int one = 1;
 gsm_option(audioCodec->priv_data, GSM_OPT_WAV49, &one);
 break;
 case CODEC_ID_GSM_MS: {
 int one = 1;
 gsm_option(audioCodec->priv_data, GSM_OPT_WAV49, &one);
 audioCodec->frame_size = 2*GSM_FRAME_SIZE;
 audioCodec->block_align = GSM_MS_BLOCK_SIZE;
 }
 }
 audioCodec->coded_frame= avcodec_alloc_frame();
 audioCodec->coded_frame->key_frame= 1;
 audioCodec->time_base = (AVRational){1, audioCodec->sample_rate};
 audioCodec->codec_type = CODEC_TYPE_AUDIO;

 if (avcodec_open(audioCodec, codec) < 0){
 __android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to avcodec_open");
 return;
 }

 bufSize = FF_MIN_BUFFER_SIZE * 10;
 buf = (uint8_t *)malloc(bufSize);
 if (buf == NULL) return;
 frameBytes = audioCodec->frame_size * audioCodec->channels * 2;
 FILE *fileWrite = fopen(FILE_NAME,"w+b");
 if(NULL == fileWrite){
 __android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to open file for reading.");
 }
 /*Write wave header*/
 WriteWav(fileWrite, 32505);/*Just for test*/

 /*Lets encode raw packet and write into file after header.*/
 __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"Lets Encode Actual Bytes");
 int nChunckSize = 0;
 while (audioSize >= frameBytes)
 {
 int packetSize;

 packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData);
 __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"Encoder returned %d bytes of data\n", packetSize);
 nChunckSize += packetSize;
 audioData += frameBytes;
 audioSize -= frameBytes;
 if(NULL != fileWrite){
 fwrite(buf, packetSize, 1, fileWrite);
 }
 else{
 __android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"Unable to open file for writting... NULL");
 }
 }
 if(NULL != fileWrite){
 fclose(fileWrite);
 }
 __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"----- Done with nChunckSize: %d --- ",nChunckSize);
 __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"*****************************");
 wavReadnDisplayHeader(FILE_NAME);
 __android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"*****************************");
 wavReadnDisplayHeader("/sdcard/Voicemail2.wav");
}



Header Writing :



/** Writes WAV headers */
void WriteWav(FILE *f, long int bytes)
{
 /* quick and dirty */
 fwrite("RIFF",sizeof(char),4,f); /* 0-3 */ //RIFF
 PutNum(bytes+44-8,f,1,4); /* 4-7 */ //ChunkSize
 fwrite("WAVEfmt ",sizeof(char),8,f); /* 8-15 */ //WAVE Header + FMT header
 PutNum(16,f,1,4); /* 16-19 */ //Size of the fmt chunk
 PutNum(49,f,1,2); /* 20-21 */ //Audio format, 49=libgsm wave, 1=PCM,6=mulaw,7=alaw, 257=IBM Mu-Law, 258=IBM A-Law, 259=ADPCM
 PutNum(1,f,1,2); /* 22-23 */ //Number of channels 1=Mono 2=Sterio
 PutNum(8000,f,1,4); /* 24-27 */ //Sampling Frequency in Hz 
 PutNum(2*8000,f,1,4); /* 28-31 */ //bytes per second /Sample/persec
 PutNum(2,f,1,2); /* 32-33 */ // 2=16-bit mono, 4=16-bit stereo 
 PutNum(16,f,1,2); /* 34-35 */ // Number of bits per sample
 fwrite("data",sizeof(char),4,f); /* 36-39 */ 
 PutNum(bytes,f,1,4); /* 40-43 */ //Sampled data length 
}



Please help....


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Wav File encoded with FFMPEG has issues with codecs while playing using VLC Player
6 décembre 2012, par user924702I want to convert raw PCM data(Taken from Android Phone mic) into a libGSM Wave file. After encoding into file, VLC player shows right codec information and duration but unable to play contents. Please help me to find what I am doing wrong.
Below is my code for encoding and header writing :
void EncodeTest(uint8_t *audioData, size_t audioSize)
{
AVCodecContext *audioCodec;
AVCodec *codec;
uint8_t *buf; int bufSize, frameBytes;
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"Lets encode :%u with size %d\n",(int)audioData, (int)audioSize);
//Set up audio encoder
codec = avcodec_find_encoder(CODEC_ID_GSM);
if (codec == NULL){
__android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to find encoder(CODEC_ID_GSM)");
codec = avcodec_find_encoder(CODEC_ID_GSM);
if (codec == NULL){
__android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to find encoder(CODEC_ID_GSM)");
return;
}
}
audioCodec = avcodec_alloc_context();
audioCodec->channels = 1;
audioCodec->sample_rate = 8000;
audioCodec->sample_fmt = SAMPLE_FMT_S16;
audioCodec->bit_rate = 13200;
audioCodec->priv_data = gsm_create();
switch(audioCodec->codec_id) {
case CODEC_ID_GSM:
audioCodec->frame_size = GSM_FRAME_SIZE;
audioCodec->block_align = GSM_BLOCK_SIZE;
int one = 1;
gsm_option(audioCodec->priv_data, GSM_OPT_WAV49, &one);
break;
case CODEC_ID_GSM_MS: {
int one = 1;
gsm_option(audioCodec->priv_data, GSM_OPT_WAV49, &one);
audioCodec->frame_size = 2*GSM_FRAME_SIZE;
audioCodec->block_align = GSM_MS_BLOCK_SIZE;
}
}
audioCodec->coded_frame= avcodec_alloc_frame();
audioCodec->coded_frame->key_frame= 1;
audioCodec->time_base = (AVRational){1, audioCodec->sample_rate};
audioCodec->codec_type = CODEC_TYPE_AUDIO;
if (avcodec_open(audioCodec, codec) < 0){
__android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to avcodec_open");
return;
}
bufSize = FF_MIN_BUFFER_SIZE * 10;
buf = (uint8_t *)malloc(bufSize);
if (buf == NULL) return;
frameBytes = audioCodec->frame_size * audioCodec->channels * 2;
FILE *fileWrite = fopen(FILE_NAME,"w+b");
if(NULL == fileWrite){
__android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"ERROR:: Unable to open file for reading.");
}
/*Write wave header*/
WriteWav(fileWrite, 32505);/*Just for test*/
/*Lets encode raw packet and write into file after header.*/
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"Lets Encode Actual Bytes");
int nChunckSize = 0;
while (audioSize >= frameBytes)
{
int packetSize;
packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData);
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"Encoder returned %d bytes of data\n", packetSize);
nChunckSize += packetSize;
audioData += frameBytes;
audioSize -= frameBytes;
if(NULL != fileWrite){
fwrite(buf, packetSize, 1, fileWrite);
}
else{
__android_log_print(ANDROID_LOG_ERROR, DEBUG_TAG,"Unable to open file for writting... NULL");
}
}
if(NULL != fileWrite){
fclose(fileWrite);
}
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"----- Done with nChunckSize: %d --- ",nChunckSize);
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"*****************************");
wavReadnDisplayHeader(FILE_NAME);
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG,"*****************************");
wavReadnDisplayHeader("/sdcard/Voicemail2.wav");
}Header Writing :
/** Writes WAV headers */
void WriteWav(FILE *f, long int bytes)
{
/* quick and dirty */
fwrite("RIFF",sizeof(char),4,f); /* 0-3 */ //RIFF
PutNum(bytesã8,f,1,4); /* 4-7 */ //ChunkSize
fwrite("WAVEfmt ",sizeof(char),8,f); /* 8-15 */ //WAVE Header + FMT header
PutNum(16,f,1,4); /* 16-19 */ //Size of the fmt chunk
PutNum(49,f,1,2); /* 20-21 */ //Audio format, 49=libgsm wave, 1=PCM,6=mulaw,7=alaw, 257=IBM Mu-Law, 258=IBM A-Law, 259=ADPCM
PutNum(1,f,1,2); /* 22-23 */ //Number of channels 1=Mono 2=Sterio
PutNum(8000,f,1,4); /* 24-27 */ //Sampling Frequency in Hz
PutNum(2*8000,f,1,4); /* 28-31 */ //bytes per second /Sample/persec
PutNum(2,f,1,2); /* 32-33 */ // 2=16-bit mono, 4=16-bit stereo
PutNum(16,f,1,2); /* 34-35 */ // Number of bits per sample
fwrite("data",sizeof(char),4,f); /* 36-39 */
PutNum(bytes,f,1,4); /* 40-43 */ //Sampled data length
}Please help....
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ffserver + ffmpeg, out rtsp stream : why ffplay cannot play the stream, but movie player can
18 juillet 2014, par user1914692Ubuntu 12.04
I use ffserver + ffmpeg.There are a little revision of the original /etc/ffserver.conf :
RTSPPort 5454
RTSPBindAddress 0.0.0.0
# ...
<stream>
Format rtp
# coming from live feed 'feed1'
Feed feed1.ffm
</stream>The command of ffmpeg (Option 1) is :
ffmpeg -re -i '/usr/share/red5/webapps/oflaDemo/streams/hobbit_vp6.flv' http://localhost:8090/feed1.ffmBTW, if I use Option 2 as below, it does not work :
ffmpeg -re -i '/usr/share/red5/webapps/oflaDemo/streams/hobbit_vp6.flv' http://192.168.1.105:8090/feed1.ffm
## not work:
[http @ 0x21a9c80] HTTP error 404 Not Found
http://192.168.1.105:8090/feed1.ffm: Input/output errorSo now I need to display the stream.
On the other computer :(Option 1)
ffplay rtsp://192.168.1.105:5454/test2-rtsp.mpgNot work. [Output] rtsp UDP timeout, retrying with TCP.
Why ?(Option 2) movie player : Open location : rtsp ://192.168.1.105:5454/test2-rtsp.mpg
it works !
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