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  • Magic Quadrant for x86 Server Virtualization Infrastructure

    - by Cinzia Mascanzoni
    The 2012 Gartner MQ for x86 Server Virtualization has just published.  KEY TAKEAWAYS - Oracle is in the “Challengers” quadrant. - This is a significant “jump” above the x-axis (from the “Niche” quadrant) during previous years - The move into the “Challengers” quadrant was possible for 3 primary reasons - 1) strength of the Oracle VM 3.0 release - 2) integrated management capabilities - 3) solid customer momentum during past year - Gartner even specifically states that Oracle VM use is growing amongst VMware customers Read the full report here.

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  • Boost Asio UDP retrieve last packet in socket buffer

    - by Alberto Toglia
    I have been messing around Boost Asio for some days now but I got stuck with this weird behavior. Please let me explain. Computer A is sending continuos udp packets every 500 ms to computer B, computer B desires to read A's packets with it own velocity but only wants A's last packet, obviously the most updated one. It has come to my attention that when I do a: mSocket.receive_from(boost::asio::buffer(mBuffer), mEndPoint); I can get OLD packets that were not processed (almost everytime). Does this make any sense? A friend of mine told me that sockets maintain a buffer of packets and therefore If I read with a lower frequency than the sender this could happen. ¡? So, the first question is how is it possible to receive the last packet and discard the ones I missed? Later I tried using the async example of the Boost documentation but found it did not do what I wanted. http://www.boost.org/doc/libs/1_36_0/doc/html/boost_asio/tutorial/tutdaytime6.html From what I could tell the async_receive_from should call the method "handle_receive" when a packet arrives, and that works for the first packet after the service was "run". If I wanted to keep listening the port I should call the async_receive_from again in the handle code. right? BUT what I found is that I start an infinite loop, it doesn't wait till the next packet, it just enters "handle_receive" again and again. I'm not doing a server application, a lot of things are going on (its a game), so my second question is, do I have to use threads to use the async receive method properly, is there some example with threads and async receive? Thanks for you attention.

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  • Packet loss rate with iperf and tcpdump

    - by stefita
    I tested a line for its link quality with iperf. The measured speed (UDP port 9005) was 96Mbps, which is fine, because both servers are connected with 100Mbps to the internet. On the other hand the datagram loss rate was shown to be 3.3-3.7%, which I found a little too much. Using a high-speed transfer protocol I recorded the packets on both sides with tcpdump. Than I calculated the packet loss - average 0.25%. Have anyone an explanation, where this big difference may be coming from? What is an acceptable packet loss in your opinion?

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  • Setting up Linux VPN Client on Mint: Never sends "Set-Link-Info" packet

    - by cabanaboy
    I have tried to set up a VPN Connection on the Linux Mint disto, but could not get it working. When I use a Windows 7 VPN client it works fine. I brought up Wireshark on both Windows and Linux machine and noticed that on the Windows machine, the client never attempted to send the "Set-Link-Info" packet whereas the Windows (working) VPN client did. Why isn't the Linux Mint client sending the "Set-Link-Info" packet. I think if it did that, then my connection would work. What am I missing?

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  • Error 2020: Got packet bigger than 'max_allowed_packet' bytes when dumping table

    - by Imagineer
    I'm getting the above mentioned error when backing up with ZRM, which is using mysqldump for backup. mysqldump --opt --extended-insert --single-transaction --create-options --default-character-set=utf8 --user=" " -p --all-databases "/nfs/backup/mysql01/dailyrun/20091216043001/backup.sql" mysqldump: Error 2020: Got packet bigger than 'max_allowed_packet' bytes when dumping table TICKET_ATTACHMENT at row: 2286 I have increased the size for 'max_allowed_packet' to be 1G in /etc/my.cnf which is the server setting and for the client side setting I've set it by running this command: mysql -u -p --max_allowed_packet=1G And I have verified that on the client and server side they are of the same value. This is to check the client side value according to this forum posting http://forums.mysql.com/read.php?35,75794,261640 mysql SELECT @@MAX_ALLOWED_PACKET - ; +----------------------+ | @@MAX_ALLOWED_PACKET | +----------------------+ | 1073741824 | +----------------------+ 1 row in set (0.00 sec) And this is the check the server value setting. mysql SHOW VARIABLES | max_allowed_packet | 1073741824 | I have ran out of ideas, and tried searching within expert exchange and googling for solutions but so far none has worked. Reference http://dev.mysql.com/doc/refman/5.1/en/packet-too-large.html Anyone please advise, thank you.

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  • How a router decides destination of packet?

    - by user58859
    I have basic networking question. Scenario : Two pc's are communicating on a wan. Both the pc's ate behind routers or modems. My question : Both the pc's have public IP of each other. That public IP is most of the time is either of the router or of the modem. There can be more then one pc's behind those routers and modems. Then how the pc's are communicating. I can understand the packets can reach upto those routers or modems. But what after that. In the packet , destination IP is public IP. Then how the router or modem decides where to send the packet? Can anybody explain me this please. Thanks in advance.

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  • How to create a magic square in PHP?

    - by TerranRich
    I'd like to try my hand at creating a Magic Square in PHP (i.e. a grid of numbers that all add up to the same value), but I really don't know where to start. I know of the many methods that create magic square, such as starting "1" at a fixed position, then moving in a specific direction with each iteration. But that doesn't create a truly randomized Magic Square, which is what I'm aiming for. I want to be able to generate an N-by-N Magic Square of N² numbers where each row and column adds up to N(N²+1)/2 (e.g. a 5x5 square where all rows/columns add up to 65 — the diagonals don't matter). Can anybody provide a starting point? I don't want anybody to do the work for me, I just need to know how to start such a project? I know of one generator, written in Java (http://www.dr-mikes-math-games-for-kids.com/how-to-make-a-magic-square.html) but the last Java experience I had was over 10 years ago before I quickly abandoned it. Therefore, I don't really understand what the code is actually doing. I did notice, however, that when you generate a new square, it shows the numbers 1-25 (for a 5x5 square), in order, before quickly generating a fresh randomized square.

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  • How does the momentum/inertial scroll work with the Magic Mouse on NSScrollView?

    - by jbrennan
    When you scroll with the newer Apple Magic Mouse (at least on 10.6, I can't confirm any previous Mac OS) you get inertial scroll like scrolling on iPhone (that is, after a flick of the finger to scroll, it doesn't abruptly stop, but instead gradually slows down). This behaviour is "for free" with all NSScrollViews, it would appear. There are exceptional cases, such as Tweetie for Mac (I've heard Tweetie was written with a custom Table View class that works akin to how UITableView works on iPhone). My question is, how do the scroll views know how to do this inertial scrolling? My guess is the mouse [driver] repeatedly sends scroll events with a dampening scroll magnitude (or something like that) over the scroll period. But I'm not really sure how it works. I am having some scrolling problems in my scrollview class and I'm trying to figure out why (obviously we don't have the source code to Tweetie to see why it doesn't get the proper scrolling), but just trying to better understand how it works in order to fix my own problems.

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  • What is the magic behind perl read() function and buffer which is not a ref ?

    - by alex8657
    I do not get to understand how the Perl read($buf) function is able to modify the content of the $buf variable. $buf is not a reference, so the parameter is given by copy (from my c/c++ knowledge). So how come the $buf variable is modified in the caller ? Is it a tie variable or something ? The C documentation about setbuf is also quite elusive and unclear to me # Example 1 $buf=''; # It is a scalar, not a ref $bytes = $fh->read($buf); print $buf; # $buf was modified, what is the magic ? # Example 2 sub read_it { my $buf = shift; return $fh->read($buf); } my $buf; $bytes = read_it($buf); print $buf; # As expected, this scope $buf was not modified

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  • WinPcap/Wireshark install: where is packet.ddl?

    - by Annonomus Penguin
    I have Wireshark installed, and I'm getting this error: The NPF driver isn't running. You may have trouble capturing or listing interfaces. I realize this is something to do with WinPcap. It's not in control panel, as the FAQ states it should be. I've tried installing it, and it says that there is a previous version installed. This leaves me to believe this is the problem: To be absolutely sure that WinPcap has been installed, please look at your system folder: you should find files called packet.* and wpcap.dll. Please check the file dates: these should be compatible with the WinPcap release dates. We've had reports of trojans or other malware that silently install the WinPcap driver, NPF.sys. If you've been infected by them, you'll probably see the driver file in Windows\System32\Drivers, but no entries in the "Add or Remove Programs" applet and no dlls. I've searched my hard drive, but the only path is this: C:\Windows\SysWOW64\packet.dll Is this the file they are talking about? Should I delete this file? I'm not quite sure, so I thought I'd verify that this file is the right one.

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  • Conceal packet loss in PCM stream

    - by ZeroDefect
    I am looking to use 'Packet Loss Concealment' to conceal lost PCM frames in an audio stream. Unfortunately, I cannot find a library that is accessible without all the licensing restrictions and code bloat (...up for some suggestions though). I have located some GPL code written by Steve Underwood for the Asterisk project which implements PLC. There are several limitations; although, as Steve suggests in his code, his algorithm can be applied to different streams with a bit of work. Currently, the code works with 8kHz 16-bit signed mono streams. Variations of the code can be found through a simple search of Google Code Search. My hope is that I can adapt the code to work with other streams. Initially, the goal is to adjust the algorithm for 8+ kHz, 16-bit signed, multichannel audio (all in a C++ environment). Eventually, I'm looking to make the code available under the GPL license in hopes that it could be of benefit to others... Attached is the code below with my efforts. The code includes a main function that will "drop" a number of frames with a given probability. Unfortunately, the code does not quite work as expected. I'm receiving EXC_BAD_ACCESS when running in gdb, but I don't get a trace from gdb when using 'bt' command. Clearly, I'm trampimg on memory some where but not sure exactly where. When I comment out the *amdf_pitch* function, the code runs without crashing... int main (int argc, char *argv[]) { std::ifstream fin("C:\\cc32kHz.pcm"); if(!fin.is_open()) { std::cout << "Failed to open input file" << std::endl; return 1; } std::ofstream fout_repaired("C:\\cc32kHz_repaired.pcm"); if(!fout_repaired.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } std::ofstream fout_lossy("C:\\cc32kHz_lossy.pcm"); if(!fout_lossy.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } audio::PcmConcealer Concealer; Concealer.Init(1, 16, 32000); //Generate random numbers; srand( time(NULL) ); int value = 0; int probability = 5; while(!fin.eof()) { char arr[2]; fin.read(arr, 2); //Generate's random number; value = rand() % 100 + 1; if(value <= probability) { char blank[2] = {0x00, 0x00}; fout_lossy.write(blank, 2); //Fill in data; Concealer.Fill((int16_t *)blank, 1); fout_repaired.write(blank, 2); } else { //Write data to file; fout_repaired.write(arr, 2); fout_lossy.write(arr, 2); Concealer.Receive((int16_t *)arr, 1); } } fin.close(); fout_repaired.close(); fout_lossy.close(); return 0; } PcmConcealer.hpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #ifndef __PCMCONCEALER_HPP__ #define __PCMCONCEALER_HPP__ /** 1. What does it do? The packet loss concealment module provides a suitable synthetic fill-in signal, to minimise the audible effect of lost packets in VoIP applications. It is not tied to any particular codec, and could be used with almost any codec which does not specify its own procedure for packet loss concealment. Where a codec specific concealment procedure exists, the algorithm is usually built around knowledge of the characteristics of the particular codec. It will, therefore, generally give better results for that particular codec than this generic concealer will. 2. How does it work? While good packets are being received, the plc_rx() routine keeps a record of the trailing section of the known speech signal. If a packet is missed, plc_fillin() is called to produce a synthetic replacement for the real speech signal. The average mean difference function (AMDF) is applied to the last known good signal, to determine its effective pitch. Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech will be repeated over and over until the real speech resumes. However, several refinements are needed to obtain smooth pleasant sounding results. - The two ends of the stored cycle of speech will not always fit together smoothly. This can cause roughness, or even clicks, at the joins between cycles. To soften this, the 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e. in total, the last 5/4 pitch periods of real speech are used). - The start of the synthetic speech will not always fit together smoothly with the tail of real speech passed on before the erasure was identified. Ideally, we would like to modify the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However, it is too late for that. We could have delayed the real speech a little, but that would require more buffer manipulation, and hurt the efficiency of the no-lost-packets case (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed, and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result seems quite acceptable. - As we progress into the erasure, the chances of the synthetic signal being anything like correct steadily fall. Therefore, the volume of the synthesized signal is made to decay linearly, such that after 50ms of missing audio it is reduced to silence. - When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the start of the real speech. If the erasure is small, this smoothes the transition. If the erasure is long, and the synthetic signal has faded to zero, the blending softens the start up of the real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. 3. How do I use it? Before audio is processed, call plc_init() to create an instance of the packet loss concealer. For each received audio packet that is acceptable (i.e. not including those being dropped for being too late) call plc_rx() to record the content of the packet. Note this may modify the packet a little after a period of packet loss, to blend real synthetic data smoothly. When a real packet is not available in time, call plc_fillin() to create a sythetic substitute. That's it! */ /*! Minimum allowed pitch (66 Hz) */ #define PLC_PITCH_MIN(SAMPLE_RATE) ((double)(SAMPLE_RATE) / 66.6) /*! Maximum allowed pitch (200 Hz) */ #define PLC_PITCH_MAX(SAMPLE_RATE) ((SAMPLE_RATE) / 200) /*! Maximum pitch OLA window */ //#define PLC_PITCH_OVERLAP_MAX(SAMPLE_RATE) ((PLC_PITCH_MIN(SAMPLE_RATE)) >> 2) /*! The length over which the AMDF function looks for similarity (20 ms) */ #define CORRELATION_SPAN(SAMPLE_RATE) ((20 * (SAMPLE_RATE)) / 1000) /*! History buffer length. The buffer must also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment. */ //#define PLC_HISTORY_LEN(SAMPLE_RATE) ((CORRELATION_SPAN(SAMPLE_RATE)) + (PLC_PITCH_MIN(SAMPLE_RATE))) namespace audio { typedef struct { /*! Consecutive erased samples */ int missing_samples; /*! Current offset into pitch period */ int pitch_offset; /*! Pitch estimate */ int pitch; /*! Buffer for a cycle of speech */ float *pitchbuf;//[PLC_PITCH_MIN]; /*! History buffer */ short *history;//[PLC_HISTORY_LEN]; /*! Current pointer into the history buffer */ int buf_ptr; } plc_state_t; class PcmConcealer { public: PcmConcealer(); ~PcmConcealer(); void Init(int channels, int bit_depth, int sample_rate); //Process a block of received audio samples. int Receive(short amp[], int frames); //Fill-in a block of missing audio samples. int Fill(short amp[], int frames); void Destroy(); private: int amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames); void save_history(plc_state_t *s, short *buf, int channel_index, int frames); void normalise_history(plc_state_t *s); /** Holds the states of each of the channels **/ std::vector< plc_state_t * > ChannelStates; int plc_pitch_min; int plc_pitch_max; int plc_pitch_overlap_max; int correlation_span; int plc_history_len; int channel_count; int sample_rate; bool Initialized; }; } #endif PcmConcealer.cpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #include "audio/PcmConcealer.hpp" /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */ #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ #if !defined(INT16_MAX) #define INT16_MAX (32767) #define INT16_MIN (-32767-1) #endif #ifdef WIN32 inline double rint(double x) { return floor(x + 0.5); } #endif inline short fsaturate(double damp) { if (damp > 32767.0) return INT16_MAX; if (damp < -32768.0) return INT16_MIN; return (short)rint(damp); } namespace audio { PcmConcealer::PcmConcealer() : Initialized(false) { } PcmConcealer::~PcmConcealer() { Destroy(); } void PcmConcealer::Init(int channels, int bit_depth, int sample_rate) { if(Initialized) return; if(channels <= 0 || bit_depth != 16) return; Initialized = true; channel_count = channels; this->sample_rate = sample_rate; ////////////// double min = PLC_PITCH_MIN(sample_rate); int imin = (int)min; double max = PLC_PITCH_MAX(sample_rate); int imax = (int)max; plc_pitch_min = imin; plc_pitch_max = imax; plc_pitch_overlap_max = (plc_pitch_min >> 2); correlation_span = CORRELATION_SPAN(sample_rate); plc_history_len = correlation_span + plc_pitch_min; ////////////// for(int i = 0; i < channel_count; i ++) { plc_state_t *t = new plc_state_t; memset(t, 0, sizeof(plc_state_t)); t->pitchbuf = new float[plc_pitch_min]; t->history = new short[plc_history_len]; ChannelStates.push_back(t); } } void PcmConcealer::Destroy() { if(!Initialized) return; while(ChannelStates.size()) { plc_state_t *s = ChannelStates.at(0); if(s) { if(s->history) delete s->history; if(s->pitchbuf) delete s->pitchbuf; memset(s, 0, sizeof(plc_state_t)); delete s; } ChannelStates.erase(ChannelStates.begin()); } ChannelStates.clear(); Initialized = false; } //Process a block of received audio samples. int PcmConcealer::Receive(short amp[], int frames) { if(!Initialized) return 0; int j = 0; for(int k = 0; k < ChannelStates.size(); k++) { int i; int overlap_len; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples) { /* Although we have a real signal, we need to smooth it to fit well with the synthetic signal we used for the previous block */ /* The start of the real data is overlapped with the next 1/4 cycle of the synthetic data. */ pitch_overlap = s->pitch >> 2; if (pitch_overlap > frames) pitch_overlap = frames; gain = 1.0 - s->missing_samples * ATTENUATION_INCREMENT; if (gain < 0.0) gain = 0.0; new_step = 1.0/pitch_overlap; old_step = new_step*gain; new_weight = new_step; old_weight = (1.0 - new_step)*gain; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[index]); if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->missing_samples = 0; } save_history(s, amp, j, frames); j++; } return frames; } //Fill-in a block of missing audio samples. int PcmConcealer::Fill(short amp[], int frames) { if(!Initialized) return 0; int j =0; for(int k = 0; k < ChannelStates.size(); k++) { short *tmp = new short[plc_pitch_overlap_max]; int i; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; short *orig_amp; int orig_len; orig_amp = amp; orig_len = frames; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples == 0) { // As the gap in real speech starts we need to assess the last known pitch, //and prepare the synthetic data we will use for fill-in normalise_history(s); s->pitch = amdf_pitch(plc_pitch_min, plc_pitch_max, s->history + plc_history_len - correlation_span - plc_pitch_min, j, correlation_span); // We overlap a 1/4 wavelength pitch_overlap = s->pitch >> 2; // Cook up a single cycle of pitch, using a single of the real signal with 1/4 //cycle OLA'ed to make the ends join up nicely // The first 3/4 of the cycle is a simple copy for (i = 0; i < s->pitch - pitch_overlap; i++) s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]; // The last 1/4 of the cycle is overlapped with the end of the previous cycle new_step = 1.0/pitch_overlap; new_weight = new_step; for ( ; i < s->pitch; i++) { s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]*(1.0 - new_weight) + s->history[plc_history_len - 2*s->pitch + i]*new_weight; new_weight += new_step; } // We should now be ready to fill in the gap with repeated, decaying cycles // of what is in pitchbuf // We need to OLA the first 1/4 wavelength of the synthetic data, to smooth // it into the previous real data. To avoid the need to introduce a delay // in the stream, reverse the last 1/4 wavelength, and OLA with that. gain = 1.0; new_step = 1.0/pitch_overlap; old_step = new_step; new_weight = new_step; old_weight = 1.0 - new_step; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->history[plc_history_len - 1 - i] + new_weight * s->pitchbuf[i]); new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->pitch_offset = i; } else { gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; i = 0; } for ( ; gain > 0.0 && i < frames; i++) { int index = (i * channel_count) + j; amp[index] = s->pitchbuf[s->pitch_offset]*gain; gain -= ATTENUATION_INCREMENT; if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; } for ( ; i < frames; i++) { int index = (i * channel_count) + j; amp[i] = 0; } s->missing_samples += orig_len; save_history(s, amp, j, frames); delete [] tmp; j++; } return frames; } void PcmConcealer::save_history(plc_state_t *s, short *buf, int channel_index, int frames) { if (frames >= plc_history_len) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ //memcpy(s->history, buf + len - plc_history_len, sizeof(short)*plc_history_len); int frames_to_copy = plc_history_len; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + frames - plc_history_len)) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = 0; return; } if (s->buf_ptr + frames > plc_history_len) { /* Wraps around - must break into two sections */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*(plc_history_len - s->buf_ptr)); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = plc_history_len - s->buf_ptr; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } frames -= (plc_history_len - s->buf_ptr); //memcpy(s->history, buf + (plc_history_len - s->buf_ptr), sizeof(short)*len); frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + (plc_history_len - s->buf_ptr))) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = frames; return; } /* Can use just one section */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*len); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } s->buf_ptr += frames; } void PcmConcealer::normalise_history(plc_state_t *s) { short *tmp = new short[plc_history_len]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(short)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(short)*(plc_history_len - s->buf_ptr)); memcpy(s->history + plc_history_len - s->buf_ptr, tmp, sizeof(short)*s->buf_ptr); s->buf_ptr = 0; delete [] tmp; } int PcmConcealer::amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames) { int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < frames; j++) { int index1 = (channel_count * (i+j)) + channel_index; int index2 = (channel_count * j) + channel_index; //std::cout << "Index 1: " << index1 << ", Index 2: " << index2 << std::endl; acc += abs(amp[index1] - amp[index2]); } if (acc < min_acc) { min_acc = acc; pitch = i; } } std::cout << "Pitch: " << pitch << std::endl; return pitch; } } P.S. - I must confess that digital audio is not my forte...

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  • Xen PV packet loss

    - by Delphinator
    I'm having some serious issues with packetloss with one of my servers. This server is a somewhat old (P4-era) machine running Debian Squeeze and Xen 4.0. There are two domUs running on it (both also Debian Squeeze), one gateway and a fileserver. Unfortunatly the processor has no virtualization extensions, therefore only PV can be used. While investigating why our network seems to be slower than it should I found some pretty bad packet loss (~25%). After further investigation and several experiments I did a measurment between the dom0 and one of the domUs: Server listening on UDP port 5001 Receiving 1470 byte datagrams UDP buffer size: 110 KByte (default) ------------------------------------------------------------ ------------------------------------------------------------ Client connecting to dom0, UDP port 5001 Sending 1470 byte datagrams UDP buffer size: 110 KByte (default) ------------------------------------------------------------ [ 3] local 192.168.1.2(domU) port 33817 connected with 192.168.1.100(dom0) port 5001 [ 4] local 192.168.1.2(domU) port 5001 connected with 192.168.1.100(dom0) port 48606 [ ID] Interval Transfer Bandwidth [ 3] 0.0-10.0 sec 46.3 MBytes 38.7 Mbits/sec [ 3] Sent 33020 datagrams [ 3] Server Report: [ 3] 0.0-10.0 sec 46.2 MBytes 38.6 Mbits/sec 0.030 ms 89/33019 (0.27%) [ 3] 0.0-10.0 sec 1 datagrams received out-of-order [ 4] 0.0-10.2 sec 43.0 MBytes 35.3 Mbits/sec 13.074 ms 11575/42256 (27%) tl;dr: 27% packet loss from dom0 to domU with 50Mbit UDP packets. Same thing happens from anywhere in the network. The problem gets better for smaller bandwidths (0.047% for 5Mbit) and worse for higher (59% for 200Mbit) ones. I did increase the CPU-weight of the dom0, there is no swapping going on, and actual networking-hardware is not involved. I never expected Xen (or anything related) to drop packets, and I'm completly clueless what to try next.

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  • Apple wireless mouse and keyboard doens't work

    - by drigoSkalWalker
    I paired the mouse and keyboard on ubuntu, but it seems not work. I got this error in /var/log/syslog kernel: [ 1875.935712] input: Apple Magic Mouse as /devices/pci0000:00/0000:00:12.0/usb4/4-3/4-3:1.0/bluetooth/hci0/hci0:2/input34 kernel: [ 1875.935885] evdev: no more free evdev devices kernel: [ 1875.935893] input: failed to attach handler evdev to device input34, error: -23 kernel: [ 1875.936049] magicmouse 0005:05AC:030D.0003: input,hidraw0: BLUETOOTH HID v3.06 Mouse [Apple Magic Mouse] on 00:19:5D:0F:4A:F6 kernel: [ 2334.787710] input: Apple Wireless Keyboard as /devices/pci0000:00/0000:00:12.0/usb4/4-3/4-3:1.0/bluetooth/hci0/hci0:4/input36 kernel: [ 2334.787729] evdev: no more free evdev devices kernel: [ 2334.787737] input: failed to attach handler evdev to device input36, error: -23 kernel: [ 2334.787999] generic-bluetooth 0005:05AC:0255.0005: input,hidraw1: BLUETOOTH HID v0.50 Keyboard [Apple Wireless Keyboard] on 00:19:5D:0F:4A:F6 Nothing appears in xinput --list, only the wired mouse and keyboard. How to fix that?

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  • sftp Bad message - (badly formatted packet or protocol incompatibility)

    - by culter
    I have two servers connected through SFTP. When I'm trying to upload file DONATE_SPLATNOSTSFRB-1503_20120315.xls.gpg via WinSCP, it works fine, but when I change file name to DONATE_SPLATNOSTSFRB-1503_20120315.gpg it sometimes upload to server and sometimes not. When It's uploaded, I have problems to delete it. I get this error message: Bad message - (badly formatted packet or protocol incompatibility) Error code: 5 Error message from server: Bad Message Request code: 13 Others files works fine e.g.: DONATE_PREDSFRB-0212_20120315.gpg Thank you.

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