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GStreamer_audioRecorder
GStreamer_audioRecorder
Commit becfa065 authored by Mario Chirinos Colunga's avatar Mario Chirinos Colunga 💬
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mario

parent 6b4a2e43
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StreamRecorder.cpp
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...@@ -19,148 +19,124 @@ ...@@ -19,148 +19,124 @@
#include <cstring> #include <cstring>
using namespace std; using namespace std;
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* Constructor * Constructor
* @param source streaing url * @param source streaing url
* @param time split recording in segments of time seconds. * @param time split recording in segments of time seconds.
*/ */
StreamRecorder::StreamRecorder(const char* source, int time)
{
int nFrames = ceil(time*STREAMRECORDER_SAMPLERATE/READSIZE);
recordTime = nFrames*READSIZE/STREAMRECORDER_SAMPLERATE;
cout << "record time: " << recordTime << endl;
StreamRecorder::StreamRecorder(const char* source, int time) { bufferSize=nFrames*READSIZE*STREAMRECORDER_BYTESPERSAMPLE;
audioBuffer = new unsigned char[bufferSize];
int nFrames = ceil(time*STREAMRECORDER_SAMPLERATE/READSIZE); memset(audioBuffer, 0, bufferSize);
recordTime = nFrames*READSIZE/STREAMRECORDER_SAMPLERATE;
cout << "record time: " << recordTime << endl;
// buffer size
bufferSize=nFrames*READSIZE*STREAMRECORDER_BYTESPERSAMPLE;
// buffer that ig going to contain the audio
audioBuffer = new unsigned char[bufferSize];
// Puts all the elements in audio buffer in 0
memset (audioBuffer, 0, bufferSize);
// set the pointer to the actual position of the buffer;
audioBufferPosition=audioBuffer;
// Bytes saved counter
nBytes=0;
// flag that save the conncetion state audioBufferPosition=audioBuffer;
isConnectionLost=false;
// Create the elements and sets capabilities nBytes=0;
createMainPipeline(); isConnectionLost=false;
// Connec with the URI createMainPipeline();
connect(source); connect(source);
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* Connect to the stream * Connect to the stream
* @param uri streaing uri * @param uri streaing uri
* @return unimplemented * @return unimplemented
*/ */
int StreamRecorder::connect(const char *uri)
{
disconnect();
cout << "connecting to " << uri << endl;
int StreamRecorder::connect(const char *uri) { pluginUri = (char*)uri;
disconnect(); gst_element_set_state (mainPipeline, GST_STATE_NULL);
cout << "connecting to " << uri << endl; gst_element_set_state (mainPipeline, GST_STATE_READY);
// Uri destiny g_object_set (G_OBJECT (streamSrc), "uri", uri, NULL);
pluginUri = (char*)uri;
gst_element_link (streamSrc, audioConvert);
gst_element_set_state (mainPipeline, GST_STATE_NULL);
gst_element_set_state (mainPipeline, GST_STATE_READY); gst_element_set_state (mainPipeline, GST_STATE_PLAYING);
return 0;
// Assign the uri property
g_object_set (G_OBJECT (streamSrc), "uri", uri, NULL);
gst_element_link (streamSrc, audioConvert);
// Main pipeline starts to work
gst_element_set_state (mainPipeline, GST_STATE_PLAYING);
return 0;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* disconnect from the stream * disconnect from the stream
* @return unimplemented * @return unimplemented
*/ */
int StreamRecorder::disconnect()
int StreamRecorder::disconnect() { {
gst_element_unlink (streamSrc, audioConvert); gst_element_unlink (streamSrc, audioConvert);
gst_element_set_state (mainPipeline, GST_STATE_NULL); gst_element_set_state (mainPipeline, GST_STATE_NULL);
return 0; return 0;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* Create main pipeline * Create main pipeline
* @return 0 on success else on error * @return 0 on success else on error
*/ */
int StreamRecorder::createMainPipeline()
{
mainPipeline = gst_pipeline_new("stream-recorder");
GstBus* bus;
bus = gst_pipeline_get_bus(GST_PIPELINE (mainPipeline));
gst_bus_add_watch(bus, bus_callback, this);
gst_object_unref(GST_OBJECT (bus));
streamSrc = gst_element_factory_make("uridecodebin", "stream_source");
audioConvert = gst_element_factory_make ("audioconvert", "audio_convert");
//audioResample = gst_element_factory_make ("audioresample", "audio_resample");
filterCaps = gst_element_factory_make("capsfilter", "filter_cap");
GstCaps *fc = gst_caps_new_full(gst_structure_new ("audio/x-raw",
"channels", G_TYPE_INT, 1,
"rate", G_TYPE_INT, STREAMRECORDER_SAMPLERATE,
"format", G_TYPE_STRING, "S16LE",
"signed", G_TYPE_BOOLEAN, 1, //MUST BE SIGNED
"endianness", G_TYPE_INT, 1234,
NULL),
NULL);
g_object_set(G_OBJECT (filterCaps), "caps", fc, NULL);
queue0 = gst_element_factory_make("queue", "queue0");
filter = gst_element_factory_make("identity", "audio_filter");
g_signal_connect(filter, "handoff", G_CALLBACK (filter_handoff_callback), this);
queue1 = gst_element_factory_make("queue", "queue1");
fakeSink = gst_element_factory_make("fakesink", "fake_sink");
//audioSink = gst_element_factory_make("autoaudiosink", "speaker");
gst_bin_add_many (GST_BIN (mainPipeline), streamSrc, audioConvert, filterCaps, queue0, filter,queue1, fakeSink, NULL);
if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, fakeSink, NULL))
{
//gst_bin_add_many (GST_BIN (mainPipeline), streamSrc, audioConvert, filterCaps, queue0, filter, queue1, audioSink, NULL);
// if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, audioSink, NULL)){
int StreamRecorder::createMainPipeline() { cerr << "mainPipeline: Failed to link elements in the pipeline" << endl;
exit(0);
mainPipeline = gst_pipeline_new("stream-recorder");
GstBus* bus;
bus = gst_pipeline_get_bus(GST_PIPELINE (mainPipeline));
gst_bus_add_watch(bus, bus_callback, this);
gst_object_unref(GST_OBJECT (bus));
streamSrc = gst_element_factory_make("uridecodebin", "stream_source");
audioConvert = gst_element_factory_make ("audioconvert", "audio_convert");
//audioResample = gst_element_factory_make ("audioresample", "audio_resample");
filterCaps = gst_element_factory_make("capsfilter", "filter_cap");
GstCaps *fc = gst_caps_new_full(gst_structure_new ("audio/x-raw",
"channels", G_TYPE_INT, 1,
"rate", G_TYPE_INT, STREAMRECORDER_SAMPLERATE,
"format", G_TYPE_STRING, "S16LE",
"signed", G_TYPE_BOOLEAN, 1, //MUST BE SIGNED
"endianness", G_TYPE_INT, 1234,
NULL),
NULL);
g_object_set(G_OBJECT (filterCaps), "caps", fc, NULL);
queue0 = gst_element_factory_make("queue", "queue0");
filter = gst_element_factory_make("identity", "audio_filter");
g_signal_connect(filter, "handoff", G_CALLBACK (filter_handoff_callback), this);
queue1 = gst_element_factory_make("queue", "queue1");
fakeSink = gst_element_factory_make("fakesink", "fake_sink");
//audioSink = gst_element_factory_make("autoaudiosink", "speaker");
gst_bin_add_many (GST_BIN (mainPipeline), streamSrc, audioConvert, filterCaps, queue0, filter,queue1, fakeSink, NULL);
if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, fakeSink, NULL)){
//gst_bin_add_many (GST_BIN (mainPipeline), streamSrc, audioConvert, filterCaps, queue0, filter, queue1, audioSink, NULL);
// if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, audioSink, NULL)){
cerr << "mainPipeline: Failed to link elements in the pipeline" << endl;
exit(0);
} }
g_signal_connect(streamSrc, "pad-added", G_CALLBACK(srcNewPad_callback), this); g_signal_connect(streamSrc, "pad-added", G_CALLBACK(srcNewPad_callback), this);
gst_element_set_state (mainPipeline, GST_STATE_NULL); gst_element_set_state (mainPipeline, GST_STATE_NULL);
return 0;
}
return 0;
}
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
gboolean StreamRecorder::reconnectURIStream(void *instance)
gboolean StreamRecorder::reconnectURIStream(void *instance){ {
cout << "Restarting the main pipeline" << endl; cout << "reconnectURIStream" << endl;
((StreamRecorder*)instance)->connect(((StreamRecorder*)instance)->pluginUri); ((StreamRecorder*)instance)->connect(((StreamRecorder*)instance)->pluginUri);
return FALSE; return FALSE;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* disconnect from the stream * disconnect from the stream
* @param the GstBus that sent the message * @param the GstBus that sent the message
...@@ -168,171 +144,147 @@ gboolean StreamRecorder::reconnectURIStream(void *instance){ ...@@ -168,171 +144,147 @@ gboolean StreamRecorder::reconnectURIStream(void *instance){
* @param user_data NULL * @param user_data NULL
* @return unimplemented * @return unimplemented
*/ */
int StreamRecorder::bus_callback (GstBus *bus, GstMessage *message, void *user_data)
int StreamRecorder::bus_callback (GstBus *bus, GstMessage *message, void *user_data) { {
//printf("StreamRecorder got %s message\n", GST_MESSAGE_TYPE_NAME (message));
//printf("StreamRecorder got %s message\n", GST_MESSAGE_TYPE_NAME (message)); if(GST_MESSAGE_TYPE (message) == GST_MESSAGE_EOS)
{
if(GST_MESSAGE_TYPE (message) == GST_MESSAGE_EOS){ ((StreamRecorder*)user_data)->isConnectionLost = true;
((StreamRecorder*)user_data)->isConnectionLost = true; }
}
switch (GST_MESSAGE_TYPE (message)) { switch (GST_MESSAGE_TYPE (message))
{
case GST_MESSAGE_EOS: case GST_MESSAGE_EOS:
cout << "End of stream" << endl; cout << "End of stream" << endl;
cout << "End sometimes src" << endl; cout << "End sometimes src" << endl;
break; break;
case GST_MESSAGE_ERROR: case GST_MESSAGE_ERROR:
gchar *debug; gchar *debug;
GError *error; GError *error;
gst_message_parse_error (message, &error, &debug); gst_message_parse_error (message, &error, &debug);
g_free (debug); g_free (debug);
cerr << "Error: "<< error->message << endl; //Print specific error
g_error_free (error);
// Print specific error g_timeout_add(60*1000, reconnectURIStream, user_data); //Try to reconnect with the uri stream
cerr << "Error: "<< error->message << endl;
g_error_free (error);
// Try to reconnect with the uri stream
g_timeout_add(60*1000, reconnectURIStream, user_data);
// Enter only if the connection is lost
if(((StreamRecorder*)user_data)->isConnectionLost){
savePartialBuffer(user_data);
((StreamRecorder*)user_data)->isConnectionLost = false;
}
if(((StreamRecorder*)user_data)->isConnectionLost) //Enter only if the connection is lost
{
savePartialBuffer(user_data);
((StreamRecorder*)user_data)->isConnectionLost = false;
}
break; break;
default: default:
break; break;
} }
return TRUE; return TRUE;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
void StreamRecorder::savePartialBuffer(void *user_data)
void StreamRecorder::savePartialBuffer(void *user_data){ {
int missingBytes = ((StreamRecorder*)user_data)->bufferSize - ((StreamRecorder*)user_data)->nBytes;
// Calculate the number of bytes that are missing for reach the max buffer size ((StreamRecorder*)user_data)->audioBufferPosition+=missingBytes;
// The buffer size is obtained based on the maximum recording time
int missingBytes = ((StreamRecorder*)user_data)->bufferSize - ((StreamRecorder*)user_data)->nBytes; ((StreamRecorder*)user_data)->compressBuffer();
((StreamRecorder*)user_data)->audioBufferPosition=((StreamRecorder*)user_data)->audioBuffer;
// Put the buffer position in the last position of the buffer memset (((StreamRecorder*)user_data)->audioBuffer, 0, ((StreamRecorder*)user_data)->bufferSize);
((StreamRecorder*)user_data)->audioBufferPosition+=missingBytes; ((StreamRecorder*)user_data)->nBytes=0;
// Save in flac format
((StreamRecorder*)user_data)->compressBuffer();
// Restart the possition pointer
((StreamRecorder*)user_data)->audioBufferPosition=((StreamRecorder*)user_data)->audioBuffer;
// Restart the buffer default values
memset (((StreamRecorder*)user_data)->audioBuffer, 0, ((StreamRecorder*)user_data)->bufferSize);
((StreamRecorder*)user_data)->nBytes=0;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* CallBack to link the pads created by uridecodebin * CallBack to link the pads created by uridecodebin
* @param element The uridecodebin element * @param element The uridecodebin element
* @param pad The pad added * @param pad The pad added
* @param data this * @param data this
*/ */
void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *data)
void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *data) { {
cout << gst_element_get_name(element)<< " adding pad.." << gst_pad_get_name (pad) << endl;
cout << "Pad Name: " << gst_pad_get_name (pad) << endl;
GstPad *sinkpad;
//gst_pad_get_caps is for gst v0.1
// GstCaps *new_pad_caps = gst_pad_get_caps (pad);
GstCaps *new_pad_caps = gst_pad_query_caps(pad, NULL); cout << gst_element_get_name(element)<< " adding pad.." << gst_pad_get_name (pad) << endl;
GstStructure *new_pad_struct = gst_caps_get_structure (new_pad_caps, 0); cout << "Pad Name: " << gst_pad_get_name (pad) << endl;
const gchar *new_pad_type = gst_structure_get_name (new_pad_struct);
GstPad *sinkpad;
//gst_pad_get_caps is for gst v0.1
// GstCaps *new_pad_caps = gst_pad_get_caps (pad);
GstCaps *new_pad_caps = gst_pad_query_caps(pad, NULL);
GstStructure *new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
const gchar *new_pad_type = gst_structure_get_name (new_pad_struct);
if (g_str_has_prefix (new_pad_type, "audio/x-raw")) { if (g_str_has_prefix (new_pad_type, "audio/x-raw"))
{
cout << "linking " << new_pad_type << endl; cout << "linking " << new_pad_type << endl;
GstElement *nextElement = ((StreamRecorder*)data)->audioConvert; GstElement *nextElement = ((StreamRecorder*)data)->audioConvert;
sinkpad = gst_element_get_static_pad (nextElement, "sink"); sinkpad = gst_element_get_static_pad (nextElement, "sink");
if (GST_PAD_LINK_FAILED (gst_pad_link (pad, sinkpad))) { if (GST_PAD_LINK_FAILED (gst_pad_link (pad, sinkpad)))
cerr << "Type is "<< new_pad_type <<" but link failed." << endl; {
exit(0); cerr << "Type is "<< new_pad_type <<" but link failed." << endl;
} exit(0);
else { }
cout <<"Link succeeded " << new_pad_type << endl; else
} cout <<"Link succeeded " << new_pad_type << endl;
} }
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* Save audio data (audioBuffer) in flac format * Save audio data (audioBuffer) in flac format
* @return unimplemented * @return unimplemented
*/ */
int StreamRecorder::compressBuffer()
int StreamRecorder::compressBuffer() { {
long int currentTime = time(NULL); long int currentTime = time(NULL);
stringstream ss; stringstream ss;
string fileNameStr, currentTimeStr; string fileNameStr, currentTimeStr;
ss << currentTime << endl; ss << currentTime << endl;
getline(ss, currentTimeStr); getline(ss, currentTimeStr);
ss << currentTimeStr << ".flac" << endl; ss << currentTimeStr << ".flac" << endl;
getline(ss, fileNameStr); getline(ss, fileNameStr);
int readsize = READSIZE; int readsize = READSIZE;
FLAC__bool ok = true; FLAC__bool ok = true;
FLAC__StreamEncoder *encoder = 0; FLAC__StreamEncoder *encoder = 0;
FLAC__StreamMetadata *metadata[2]; FLAC__StreamMetadata *metadata[2];
FLAC__StreamMetadata_VorbisComment_Entry entry; FLAC__StreamMetadata_VorbisComment_Entry entry;
FLAC__StreamEncoderInitStatus init_status; FLAC__StreamEncoderInitStatus init_status;
/* allocate the encoder */ /* allocate the encoder */
if((encoder = FLAC__stream_encoder_new()) == NULL) { if((encoder = FLAC__stream_encoder_new()) == NULL)
fprintf(stderr, "ERROR: allocating encoder\n"); {
return 1; fprintf(stderr, "ERROR: allocating encoder\n");
return 1;
} }
ok &= FLAC__stream_encoder_set_verify(encoder, true); ok &= FLAC__stream_encoder_set_verify(encoder, true);
ok &= FLAC__stream_encoder_set_compression_level(encoder, 5); ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
ok &= FLAC__stream_encoder_set_channels(encoder, 1); ok &= FLAC__stream_encoder_set_channels(encoder, 1);
ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, STREAMRECORDER_BYTESPERSAMPLE*8); ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, STREAMRECORDER_BYTESPERSAMPLE*8);
ok &= FLAC__stream_encoder_set_sample_rate(encoder, STREAMRECORDER_SAMPLERATE); ok &= FLAC__stream_encoder_set_sample_rate(encoder, STREAMRECORDER_SAMPLERATE);
ok &= FLAC__stream_encoder_set_total_samples_estimate(encoder, nBytes/STREAMRECORDER_BYTESPERSAMPLE); ok &= FLAC__stream_encoder_set_total_samples_estimate(encoder, nBytes/STREAMRECORDER_BYTESPERSAMPLE);
/* now add some metadata; we'll add some tags and a padding block */ /* now add some metadata; we'll add some tags and a padding block */
if(ok) { if(ok)
if( {
(metadata[0] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) == NULL || if(
(metadata[1] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PADDING)) == NULL || (metadata[0] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) == NULL ||
/* there are many tag (vorbiscomment) functions but these are convenient for this particular use: */ (metadata[1] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PADDING)) == NULL ||
!FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "ARTIST", "artist") || /* there are many tag (vorbiscomment) functions but these are convenient for this particular use: */
!FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, /*copy=*/false) || /* copy=false: let metadata object take control of entry's allocated string */ !FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "ARTIST", "artist") ||
!FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "YEAR", "year") || !FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, /*copy=*/false) || /* copy=false: let metadata object take control of entry's allocated string */
!FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, /*copy=*/false) !FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&entry, "YEAR", "year") ||
) !FLAC__metadata_object_vorbiscomment_append_comment(metadata[0], entry, /*copy=*/false)
)
{ {
fprintf(stderr, "ERROR: out of memory or tag error\n"); fprintf(stderr, "ERROR: out of memory or tag error\n");
ok = false; ok = false;
} }
metadata[1]->length = 1234; /* set the padding length */ metadata[1]->length = 1234; /* set the padding length */
ok = FLAC__stream_encoder_set_metadata(encoder, metadata, 2); ok = FLAC__stream_encoder_set_metadata(encoder, metadata, 2);
} }
...@@ -347,87 +299,84 @@ int StreamRecorder::compressBuffer() { ...@@ -347,87 +299,84 @@ int StreamRecorder::compressBuffer() {
} }
} }
int channels = 1; int channels = 1;
// int bps = 16; // int bps = 16;
unsigned char* audioBufferTmp = audioBuffer; unsigned char* audioBufferTmp = audioBuffer;
unsigned char* buffer = new unsigned char[readsize*STREAMRECORDER_BYTESPERSAMPLE*channels]; unsigned char* buffer = new unsigned char[readsize*STREAMRECORDER_BYTESPERSAMPLE*channels];
int* pcm = new int[readsize*channels]; int* pcm = new int[readsize*channels];
/* read blocks of samples from WAVE file and feed to encoder */ /* read blocks of samples from WAVE file and feed to encoder */
if(ok) { if(ok)
int left = nBytes; {
int left = nBytes;
while(left&&ok) { while(left&&ok)
{
int need = (left>readsize? readsize : left);
memcpy(buffer, audioBufferTmp, need);//*sizeof(short)); int need = (left>readsize? readsize : left);
audioBufferTmp+=need;//*sizeof(short); memcpy(buffer, audioBufferTmp, need);//*sizeof(short));
// /* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */ audioBufferTmp+=need;//*sizeof(short);
for(int i=0; i < need*channels; i++) { // /* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
/* inefficient but simple and works on big- or little-endian machines */ for(int i=0; i < need*channels; i++)
pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]); {
/* inefficient but simple and works on big- or little-endian machines */
pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]);
} }
/* feed samples to encoder */ /* feed samples to encoder */
ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need/STREAMRECORDER_BYTESPERSAMPLE); ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need/STREAMRECORDER_BYTESPERSAMPLE);
left -= need; left -= need;
} }
} }
else { else
cout << "-ERROR-" << endl; cout << "-ERROR-" << endl;
}
ok &= FLAC__stream_encoder_finish(encoder);
fprintf(stderr, "encoding: %s\n", ok? "succeeded" : "FAILED");
fprintf(stderr, " state: %s\n", FLAC__StreamEncoderStateString[FLAC__stream_encoder_get_state(encoder)]);
/* now that encoding is finished, the metadata can be freed */
FLAC__metadata_object_delete(metadata[0]);
FLAC__metadata_object_delete(metadata[1]);
FLAC__stream_encoder_delete(encoder);
delete[] buffer; ok &= FLAC__stream_encoder_finish(encoder);
delete[] pcm;
return 0; fprintf(stderr, "encoding: %s\n", ok? "succeeded" : "FAILED");
fprintf(stderr, " state: %s\n", FLAC__StreamEncoderStateString[FLAC__stream_encoder_get_state(encoder)]);
/* now that encoding is finished, the metadata can be freed */
FLAC__metadata_object_delete(metadata[0]);
FLAC__metadata_object_delete(metadata[1]);
FLAC__stream_encoder_delete(encoder);
delete[] buffer;
delete[] pcm;
return 0;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* Add audio data to audioBuffer * Add audio data to audioBuffer
* @param data Audio data to add * @param data Audio data to add
* @param length Data length * @param length Data length
* @return Bytes writen * @return Bytes writen
*/ */
int StreamRecorder::addToBuffer(unsigned char* data, int length) { int StreamRecorder::addToBuffer(unsigned char* data, int length) {
//cout << "addToBuffer(" << length << ")" << endl; //cout << "addToBuffer(" << length << ")" << endl;
int bytesRead = length;// READSIZE*STREAMRECORDER_BYTESPERSAMPLE; int bytesRead = length;// READSIZE*STREAMRECORDER_BYTESPERSAMPLE;
memcpy((char*)audioBufferPosition, (char*)data, bytesRead); memcpy((char*)audioBufferPosition, (char*)data, bytesRead);
cout << *audioBufferPosition << endl; cout << *audioBufferPosition << endl;
audioBufferPosition+=bytesRead; audioBufferPosition+=bytesRead;
nBytes+=bytesRead;//READSIZE; nBytes+=bytesRead;//READSIZE;
cout << "Bytes readed " << nBytes << endl; cout << "Bytes readed " << nBytes << endl;
cout << "Buffer size " << bufferSize << endl; cout << "Buffer size " << bufferSize << endl;
if(nBytes >= bufferSize) { if(nBytes >= bufferSize)
compressBuffer(); {
audioBufferPosition=audioBuffer; compressBuffer();
memset (audioBuffer, 0, bufferSize); audioBufferPosition=audioBuffer;
nBytes=0; memset (audioBuffer, 0, bufferSize);
} nBytes=0;
}
return nBytes; return nBytes;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** /**
* CallBack for handoff signal of identity filter * CallBack for handoff signal of identity filter
* @param filter Identity filter * @param filter Identity filter
...@@ -435,22 +384,23 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) { ...@@ -435,22 +384,23 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) {
* @param user_data this * @param user_data this
* @return unimplemented * @return unimplemented
*/ */
int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffer, void* user_data)
int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffer, void* user_data) { {
cout << "LLEGO INFO" << endl;
GstMapInfo info;
if(!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
cout << "ERROR: MAPPING IS NOT VALID" << endl;
}
//GST_BUFFER_DATA is for gst v0.1 cout << "LLEGO INFO" << endl;
// ((StreamRecorder*)user_data)->addToBuffer((unsigned char*)GST_BUFFER_DATA (buffer)); GstMapInfo info;
if(!gst_buffer_map (buffer, &info, GST_MAP_READ))
{
cout << "ERROR: MAPPING IS NOT VALID" << endl;
}
// user data is the class //GST_BUFFER_DATA is for gst v0.1
((StreamRecorder*)user_data)->addToBuffer((unsigned char*)info.data, info.size); // ((StreamRecorder*)user_data)->addToBuffer((unsigned char*)GST_BUFFER_DATA (buffer));
gst_buffer_unmap (buffer, &info);
return 0; // user data is the class
((StreamRecorder*)user_data)->addToBuffer((unsigned char*)info.data, info.size);
gst_buffer_unmap (buffer, &info);
return 0;
} }
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
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