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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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  • PC BluRay - Multichannel HD Audio output

    - by sheepsimulator
    When playing a BluRay movie on a PC (any OS, Mac/Win/Linux), I have some questions about audio output: When playing a BluRay disc on the PC using a BluRay player program, can it decode the multichannel (7.1) LPCM/ Dolby Digital Plus / Dolby TrueHD / DTS-HD / DTS-HDMA soundtracks in their HD formats (ie, without downmixing to Dolby Digital or DTS or PCM) and output the audio directly to the soundcard's 7.1 line-level analog outputs? Is it possible to bitstream the the multichannel (7.1) LPCM/ Dolby Digital Plus / Dolby TrueHD / DTS-HD / DTS-HDMA soundtracks in their HD formats (ie, without downmixing to Dolby Digital or DTS or PCM) over the HDMI output to a receiver when using a BluRay player program? I'd kinda like to know. I'm contemplating building a home theater PC, and the above functionality is important. I'd prefer that #1 is possible, actually, because it would mean I wouldn't have to buy a receiver.

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  • Laptop wakes from sleep, once, due to audio controller (Windows 7)

    - by stijn
    The laptop is a recent Dell XPS 15z and the problem is as follows (reproducible about 90% of tries): put laptop to sleep using either Start-Sleep or closing the lid laptop goes to sleep after about 5 seconds, but instantly wakes again showing a black screen (touching the keyboard or moving the mouse shows the login screen one normally gets after wake) login again, put laptop to sleep latop stays in sleep mode output of powercfg -lastwake after the first instant wake shows the audio controller is responsible. Why would that be, why only the first try, and how to fix this? Wake History Count - 1 Wake History [0] Wake Source Count - 1 Wake Source [0] Type: Device Instance Path: PCI\VEN_8086&DEV_1C20&SUBSYS_04461028&REV_05\3&11583659&0&D8 Friendly Name: Description: High Definition Audio Controller Manufacturer: Microsoft

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  • FFmpeg audio dont work in converted videos

    - by Juddy Swaft
    NOTICE: when i convert videos via terminal and download them from ftp into pc the audio works fine. I use: if($ext == "avi" && $convert_avi == true) { $convert_source = _VIDEOS_DIR_PATH.$new_name; $conv_name = substr(md5($file['name'].rand(1,888)), 2, 10).".mp4"; $converted_file = _VIDEOS_DIR_PATH.$conv_name; $ffmpeg_command = 'ffmpeg -i '.$convert_source.' -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 '.$converted_file; echo exec($ffmpeg_command); $sql = "UPDATE pm_temp SET url = '".$conv_name."' WHERE url = '".$new_name."' LIMIT 1"; $result = @mysql_query($sql); unlink($convert_source); } This code to convert avi to mp4 ffmpeg concole output: root@1tb:~# ffmpeg -i sample.avi -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 goodsample.mp4 ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers built on Feb 22 2013 07:18:58 with gcc 4.4.5 configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libfreetype --enable-libschroedinger --disable-encoder=libschroedinger - s libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.123. 0 / 52.123. 0 libavformat 52.111. 0 / 52.111. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 [mp3 @ 0x191d4100] Header missing [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'sample.avi': Metadata: encoder : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:01:01.81, start: 0.000000, bitrate: 1194 kb/s Stream #0.0: Video: mpeg4, yuv420p, 640x352 [PAR 1:1 DAR 20:11], 23.98 tbr, Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s [buffer @ 0x191d1c80] w:640 h:352 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x191d6880] w:640 h:352 fmt:yuv420p -> w:1280 h:720 fmt:yuv420p flags:0 [libx264 @ 0x191ce5a0] Default settings detected, using medium profile [libx264 @ 0x191ce5a0] using SAR=45/44 [libx264 @ 0x191ce5a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle S [libx264 @ 0x191ce5a0] profile High, level 3.1 [libx264 @ 0x191ce5a0] 264 - core 118 - H.264/MPEG-4 AVC codec - Copyleft 2003-2 6 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_off 1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_l Output #0, mp4, to 'goodsample.mp4': Metadata: encoder : Lavf52.111.0 Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 45:44 DAR 20:11], q=2-31 Stream #0.1: Audio: libmp3lame, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help [mp3 @ 0x191d4100] Header missing Error while decoding stream #0.1 [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected [mp3 @ 0x191d4100] incomplete frame 9467kB time=00:01:00.32 bitrate=1285.5kbits/ Error while decoding stream #0.1 frame= 1852 fps= 20 q=29.0 Lsize= 9652kB time=00:01:01.72 bitrate=1280.9kbits video:9121kB audio:483kB global headers:0kB muxing overhead 0.499688% frame I:11 Avg QP:16.78 size: 51456 [libx264 @ 0x191ce5a0] frame P:784 Avg QP:20.81 size: 8954 [libx264 @ 0x191ce5a0] frame B:1057 Avg QP:26.06 size: 1659 [libx264 @ 0x191ce5a0] consecutive B-frames: 22.0% 3.1% 7.5% 67.4% [libx264 @ 0x191ce5a0] mb I I16..4: 31.1% 59.8% 9.1% [libx264 @ 0x191ce5a0] mb P I16..4: 1.8% 2.6% 0.2% P16..4: 24.3% 7.0% 4.0 [libx264 @ 0x191ce5a0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 22.7% 0.8% 0.2 [libx264 @ 0x191ce5a0] 8x8 transform intra:57.0% inter:72.6% [libx264 @ 0x191ce5a0] coded y,uvDC,uvAC intra: 44.4% 33.3% 10.3% inter: 7.6% 5. [libx264 @ 0x191ce5a0] i16 v,h,dc,p: 68% 14% 8% 10% [libx264 @ 0x191ce5a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 27% 5% 7% 7% 6 [libx264 @ 0x191ce5a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 14% 14% 6% 10% 9% 7 [libx264 @ 0x191ce5a0] i8c dc,h,v,p: 67% 13% 17% 3% [libx264 @ 0x191ce5a0] Weighted P-Frames: Y:1.9% UV:0.4% [libx264 @ 0x191ce5a0] ref P L0: 62.2% 12.8% 10.3% 14.5% 0.2% [libx264 @ 0x191ce5a0] ref B L0: 88.1% 5.5% 6.4% [libx264 @ 0x191ce5a0] ref B L1: 95.7% 4.3% [libx264 @ 0x191ce5a0] kb/s:1209.03 I know there is couple errors tough, but i dont know hot to fix it. Also i would be very thankfull if someone can help reduce video size but is not main problem video weights as original avi but sill.

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  • Use all 5.1 speakers with a 2.1 audio source

    - by thegreyspot
    Hi! I just bought a 5.1 surround sound speaker set for my computer in my bedroom. The rear speakers are next to me in bed while the front speakers are at the other end of the bed at my feet. While I enjoy the surround sound during movies that support 5.1 sound, I would like to have my rear speakers working when listening to podcasts, or other 2.1 channel sound. How can I do this? When I enable "Speaker Fill" in the Realtek Hd Audio manager the sound only comes out of the front and center speakers with a few background noises that come out the rear ones. But since my ears are closer to the rear speakers, I'd rather have the sound come out of them. Let me know of any ideas! Hmm seems like the only option is to set the rear speakers to "Front Speakers" and change it to stereo in the Realtek HD audio. But still that take alot of steps and it doesnt not use the center speaker Thanks

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  • Realtek HD Audio playing weird with certain video formats

    - by dyasny
    Hi, I have a Gigabyte motherboard with an onboard Realtek HD sound card. The card is working perfectly everywhere, except for a single video format, where the voice is distorted, sounds as if it's been passed through a metal tube. Been googling for this, but couldn't find an answer anywhere. The movie plays fine on other systems (got Linux everywhere else), but on this one (winXP-x64-sp2) it just doesn't. Here are some details: MPC: Type: KLCP WMV File Audio: 0x000a 22050Hz mono 20Kbps [Raw Audio 0] Video: Windows Media Video 9 400x300 29.97fps 227Kbps [Raw Video 1] VLC: Codec: wmas Sample rate: 22050 Bits per sample: 16 Bitrate: 20kb/s

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  • In search of a good audio player for Ubuntu 9.10

    - by Joe Casadonte
    If this should be marked Community Wiki, please let me know. I'm switching from XP to Ubuntu, and I have been very disappointed with the selection of media players available. I'm primarily interested in an audio player, but integrated video and library management is OK, too. My criteria: Must be able to play audio CDs (I'm shocked how many apps this does away with, right away) Must be able to play MP3 & WAV; OGG, SHN, FLAC are all bonuses Repeat and Shuffle modes are a must FreeDB / GraceNote through a proxy is a must (if it can read a PAC file, that would be awesome) It needs to be really small, e.g. skinnable or an applet Ability to execute a playlist is a plus Gapless MP3 playback a plus I'm running Gnome, but I'm not totally adverse to a KDE app. Command-line only is also a viable option. Some that I've tried: RhythmBox - probably the best of the lot that I've tried; I don't like its mini mode (doesn't show the song being played) and I can't figure out how to get it to hit FreeDB/GraceNote through a proxy Songbird - can't play CDs, playlist management is atrocious Banshee Jajuk Maybe a couple of more. Thanks! UPDATE I tried out VLC, Amarok and Songbord (again). VLC I eventually got to work (I had some kind of bad configuration). It seemed way more involved than I was looking for out of a music player, and in general more geared to video than audio. I couldn't fathom its library management, which I think it has; maybe it doesn't, and that's why I couldn't figure it out. Amaork looked very promising but the library management was not to my liking, and the way it handled a playlist with both MP3 and WAV is inexplicable at best. I did like some aspects of the UI, but not enough to keep it. Songbird is very finicky, but I like the library management. Sort of. It kept telling me my Watch folder was invalid, even thought it clearly was accessible. Playlist management is bizarre, and the message that it was deleting source files whenever I deleted a playlist had me too worried to keep using it. Had it been able to play CDs, maybe I would have persevered. Audacious, while a bit odd at times, does seem to do what I want. If it had a library manager, I wouldn't have bothered trying any of the others. Thanks for the help, everyone!

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  • Creating video with audio and still image for YouTube

    - by scottlabs
    I'm running the following command: ffmpeg -i audio.mp3 -ar 44100 -f image2 -i logo.jpg -r 15 -b 1800 -s 640x480 foo.mov Which successfully outputs a video with my recorded audio and an image on it. When I try and upload this to YouTube it fails to process, regardless of the formats I try: .mov, .avi, .flv, .mp4 Is there some setting I'm missing in the above that would generate a format Youtube will accept? I've tried looking through the ffmpeg documentation but I'm in over my head. I did an experiment by putting a 2 second video with a 30 second mp3. When I uploaded to youtube, the resulting video was only 2 seconds long. So it may be that YouTube looks only to the video track for the length, and since a picture is only a frame long or whatever, maybe that borks it.

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  • Can extension methods be applied to interfaces?

    - by Greg
    Hi, Is it possible to apply an extension method to an interface? (C# question) That is for example to achieve the following: create an ITopology interface create an extension method for this interface (e.g. public static int CountNodes(this ITopology topologyIf) ) then when creating a class (e.g. MyGraph) which implements ITopology, then it would automatically have the Count Nodes extension. This way the classes implementing the interface would not have to have a set class name to align with what was defined in the extension method.

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  • Items Created via IB are not being displayed

    - by Rohan
    Hello, This is what i have done so far, Made a Custom view added few objects, as shown below @interface CommUICustomSignInView : CommUICustomView { IBOutlet NSButton *pBtn; IBOutlet NSTextField *pTextField; NSTrackingArea *pTrackingArea; NSCursor *pPonitHandCursor; } Added all IBoutlet correctly, Now thru Interface builder constructed a Window controller interface builder in which added a custom view and provided this class name in the interface builder, I hope i could able to visualize you guys, now whenever i loaded this view, the item whatever i created by IB are not being displayed, Am i doing something wrong, or something missing,

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  • Simple Cisco ASA 5505 config issue

    - by Ben Sebborn
    I have a Cisco ASA setup with two interfaces: inside: 192.168.2.254 / 255.255.255.0 SecLevel:100 outside: 192.168.3.250 / 255.255.255.0 SecLevel: 0 I have a static route setup to allow PCs on the inside network to access the internet via a gateway on the outside interface (3.254): outside 0.0.0.0 0.0.0.0 192.168.3.254 This all works fine. I now need to be able to access a PC on the outside interface (3.253) from a PC on the inside interface on port 35300. I understand I should be able to do this with no problems, as I'm going from a higher security level to a lower one. However I can't get any connection. Do I need to set up a seperate static route? Perhaps the route above is overriding what I need to be able to do (is it routing ALL traffic through the gateway?) Any advice on how to do this would be apprecaited. I am configuring this via ASDM but the config can be seen as below: Result of the command: "show running-config" : Saved : ASA Version 8.2(5) ! hostname ciscoasa domain-name xxx.internal names name 192.168.2.201 dev.xxx.internal description Internal Dev server name 192.168.2.200 Newserver ! interface Ethernet0/0 switchport access vlan 2 ! interface Ethernet0/1 ! interface Ethernet0/2 ! interface Ethernet0/3 shutdown ! interface Ethernet0/4 shutdown ! interface Ethernet0/5 shutdown ! interface Ethernet0/6 shutdown ! interface Ethernet0/7 shutdown ! interface Vlan1 nameif inside security-level 100 ip address 192.168.2.254 255.255.255.0 ! interface Vlan2 nameif outside security-level 0 ip address 192.168.3.250 255.255.255.0 ! ! time-range Workingtime periodic weekdays 9:00 to 18:00 ! ftp mode passive clock timezone GMT/BST 0 clock summer-time GMT/BDT recurring last Sun Mar 1:00 last Sun Oct 2:00 dns domain-lookup inside dns server-group DefaultDNS name-server Newserver domain-name xxx.internal same-security-traffic permit inter-interface object-group service Mysql tcp port-object eq 3306 object-group protocol TCPUDP protocol-object udp protocol-object tcp access-list inside_access_in extended permit ip any any access-list outside_access_in remark ENABLES OUTSDIE ACCESS TO DEV SERVER! access-list outside_access_in extended permit tcp any interface outside eq www time-range Workingtime inactive access-list outside_access_in extended permit tcp host www-1.xxx.com interface outside eq ssh access-list inside_access_in_1 extended permit tcp any any eq www access-list inside_access_in_1 extended permit tcp any any eq https access-list inside_access_in_1 remark Connect to SSH services access-list inside_access_in_1 extended permit tcp any any eq ssh access-list inside_access_in_1 remark Connect to mysql server access-list inside_access_in_1 extended permit tcp any host mysql.xxx.com object-group Mysql access-list inside_access_in_1 extended permit tcp any host mysql.xxx.com eq 3312 access-list inside_access_in_1 extended permit object-group TCPUDP host Newserver any eq domain access-list inside_access_in_1 extended permit icmp any any access-list inside_access_in_1 remark Draytek Admin access-list inside_access_in_1 extended permit tcp any 192.168.3.0 255.255.255.0 eq 4433 access-list inside_access_in_1 remark Phone System access-list inside_access_in_1 extended permit tcp any 192.168.3.0 255.255.255.0 eq 35300 log disable pager lines 24 logging enable logging asdm warnings logging from-address [email protected] logging recipient-address [email protected] level errors mtu inside 1500 mtu outside 1500 ip verify reverse-path interface inside ip verify reverse-path interface outside ipv6 access-list inside_access_ipv6_in permit tcp any any eq www ipv6 access-list inside_access_ipv6_in permit tcp any any eq https ipv6 access-list inside_access_ipv6_in permit tcp any any eq ssh ipv6 access-list inside_access_ipv6_in permit icmp6 any any icmp unreachable rate-limit 1 burst-size 1 icmp permit any outside no asdm history enable arp timeout 14400 global (outside) 1 interface nat (inside) 1 0.0.0.0 0.0.0.0 static (inside,outside) tcp interface www dev.xxx.internal www netmask 255.255.255.255 static (inside,outside) tcp interface ssh dev.xxx.internal ssh netmask 255.255.255.255 access-group inside_access_in in interface inside control-plane access-group inside_access_in_1 in interface inside access-group inside_access_ipv6_in in interface inside access-group outside_access_in in interface outside route outside 0.0.0.0 0.0.0.0 192.168.3.254 10 route outside 192.168.3.252 255.255.255.255 192.168.3.252 1 timeout xlate 3:00:00 timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02 timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00 timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00 timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute timeout tcp-proxy-reassembly 0:01:00 timeout floating-conn 0:00:00 dynamic-access-policy-record DfltAccessPolicy aaa authentication telnet console LOCAL aaa authentication enable console LOCAL

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  • Creating a dynamic proxy generator – Part 1 – Creating the Assembly builder, Module builder and cach

    - by SeanMcAlinden
    I’ve recently started a project with a few mates to learn the ins and outs of Dependency Injection, AOP and a number of other pretty crucial patterns of development as we’ve all been using these patterns for a while but have relied totally on third part solutions to do the magic. We thought it would be interesting to really get into the details by rolling our own IoC container and hopefully learn a lot on the way, and you never know, we might even create an excellent framework. The open source project is called Rapid IoC and is hosted at http://rapidioc.codeplex.com/ One of the most interesting tasks for me is creating the dynamic proxy generator for enabling Aspect Orientated Programming (AOP). In this series of articles, I’m going to track each step I take for creating the dynamic proxy generator and I’ll try my best to explain what everything means - mainly as I’ll be using Reflection.Emit to emit a fair amount of intermediate language code (IL) to create the proxy types at runtime which can be a little taxing to read. It’s worth noting that building the proxy is without a doubt going to be slightly painful so I imagine there will be plenty of areas I’ll need to change along the way. Anyway lets get started…   Part 1 - Creating the Assembly builder, Module builder and caching mechanism Part 1 is going to be a really nice simple start, I’m just going to start by creating the assembly, module and type caches. The reason we need to create caches for the assembly, module and types is simply to save the overhead of recreating proxy types that have already been generated, this will be one of the important steps to ensure that the framework is fast… kind of important as we’re calling the IoC container ‘Rapid’ – will be a little bit embarrassing if we manage to create the slowest framework. The Assembly builder The assembly builder is what is used to create an assembly at runtime, we’re going to have two overloads, one will be for the actual use of the proxy generator, the other will be mainly for testing purposes as it will also save the assembly so we can use Reflector to examine the code that has been created. Here’s the code: DynamicAssemblyBuilder using System; using System.Reflection; using System.Reflection.Emit; namespace Rapid.DynamicProxy.Assembly {     /// <summary>     /// Class for creating an assembly builder.     /// </summary>     internal static class DynamicAssemblyBuilder     {         #region Create           /// <summary>         /// Creates an assembly builder.         /// </summary>         /// <param name="assemblyName">Name of the assembly.</param>         public static AssemblyBuilder Create(string assemblyName)         {             AssemblyName name = new AssemblyName(assemblyName);               AssemblyBuilder assembly = AppDomain.CurrentDomain.DefineDynamicAssembly(                     name, AssemblyBuilderAccess.Run);               DynamicAssemblyCache.Add(assembly);               return assembly;         }           /// <summary>         /// Creates an assembly builder and saves the assembly to the passed in location.         /// </summary>         /// <param name="assemblyName">Name of the assembly.</param>         /// <param name="filePath">The file path.</param>         public static AssemblyBuilder Create(string assemblyName, string filePath)         {             AssemblyName name = new AssemblyName(assemblyName);               AssemblyBuilder assembly = AppDomain.CurrentDomain.DefineDynamicAssembly(                     name, AssemblyBuilderAccess.RunAndSave, filePath);               DynamicAssemblyCache.Add(assembly);               return assembly;         }           #endregion     } }   So hopefully the above class is fairly explanatory, an AssemblyName is created using the passed in string for the actual name of the assembly. An AssemblyBuilder is then constructed with the current AppDomain and depending on the overload used, it is either just run in the current context or it is set up ready for saving. It is then added to the cache.   DynamicAssemblyCache using System.Reflection.Emit; using Rapid.DynamicProxy.Exceptions; using Rapid.DynamicProxy.Resources.Exceptions;   namespace Rapid.DynamicProxy.Assembly {     /// <summary>     /// Cache for storing the dynamic assembly builder.     /// </summary>     internal static class DynamicAssemblyCache     {         #region Declarations           private static object syncRoot = new object();         internal static AssemblyBuilder Cache = null;           #endregion           #region Adds a dynamic assembly to the cache.           /// <summary>         /// Adds a dynamic assembly builder to the cache.         /// </summary>         /// <param name="assemblyBuilder">The assembly builder.</param>         public static void Add(AssemblyBuilder assemblyBuilder)         {             lock (syncRoot)             {                 Cache = assemblyBuilder;             }         }           #endregion           #region Gets the cached assembly                  /// <summary>         /// Gets the cached assembly builder.         /// </summary>         /// <returns></returns>         public static AssemblyBuilder Get         {             get             {                 lock (syncRoot)                 {                     if (Cache != null)                     {                         return Cache;                     }                 }                   throw new RapidDynamicProxyAssertionException(AssertionResources.NoAssemblyInCache);             }         }           #endregion     } } The cache is simply a static property that will store the AssemblyBuilder (I know it’s a little weird that I’ve made it public, this is for testing purposes, I know that’s a bad excuse but hey…) There are two methods for using the cache – Add and Get, these just provide thread safe access to the cache.   The Module Builder The module builder is required as the create proxy classes will need to live inside a module within the assembly. Here’s the code: DynamicModuleBuilder using System.Reflection.Emit; using Rapid.DynamicProxy.Assembly; namespace Rapid.DynamicProxy.Module {     /// <summary>     /// Class for creating a module builder.     /// </summary>     internal static class DynamicModuleBuilder     {         /// <summary>         /// Creates a module builder using the cached assembly.         /// </summary>         public static ModuleBuilder Create()         {             string assemblyName = DynamicAssemblyCache.Get.GetName().Name;               ModuleBuilder moduleBuilder = DynamicAssemblyCache.Get.DefineDynamicModule                 (assemblyName, string.Format("{0}.dll", assemblyName));               DynamicModuleCache.Add(moduleBuilder);               return moduleBuilder;         }     } } As you can see, the module builder is created on the assembly that lives in the DynamicAssemblyCache, the module is given the assembly name and also a string representing the filename if the assembly is to be saved. It is then added to the DynamicModuleCache. DynamicModuleCache using System.Reflection.Emit; using Rapid.DynamicProxy.Exceptions; using Rapid.DynamicProxy.Resources.Exceptions; namespace Rapid.DynamicProxy.Module {     /// <summary>     /// Class for storing the module builder.     /// </summary>     internal static class DynamicModuleCache     {         #region Declarations           private static object syncRoot = new object();         internal static ModuleBuilder Cache = null;           #endregion           #region Add           /// <summary>         /// Adds a dynamic module builder to the cache.         /// </summary>         /// <param name="moduleBuilder">The module builder.</param>         public static void Add(ModuleBuilder moduleBuilder)         {             lock (syncRoot)             {                 Cache = moduleBuilder;             }         }           #endregion           #region Get           /// <summary>         /// Gets the cached module builder.         /// </summary>         /// <returns></returns>         public static ModuleBuilder Get         {             get             {                 lock (syncRoot)                 {                     if (Cache != null)                     {                         return Cache;                     }                 }                   throw new RapidDynamicProxyAssertionException(AssertionResources.NoModuleInCache);             }         }           #endregion     } }   The DynamicModuleCache is very similar to the assembly cache, it is simply a statically stored module with thread safe Add and Get methods.   The DynamicTypeCache To end off this post, I’m going to create the cache for storing the generated proxy classes. I’ve spent a fair amount of time thinking about the type of collection I should use to store the types and have finally decided that for the time being I’m going to use a generic dictionary. This may change when I can actually performance test the proxy generator but the time being I think it makes good sense in theory, mainly as it pretty much maintains it’s performance with varying numbers of items – almost constant (0)1. Plus I won’t ever need to loop through the items which is not the dictionaries strong point. Here’s the code as it currently stands: DynamicTypeCache using System; using System.Collections.Generic; using System.Security.Cryptography; using System.Text; namespace Rapid.DynamicProxy.Types {     /// <summary>     /// Cache for storing proxy types.     /// </summary>     internal static class DynamicTypeCache     {         #region Declarations           static object syncRoot = new object();         public static Dictionary<string, Type> Cache = new Dictionary<string, Type>();           #endregion           /// <summary>         /// Adds a proxy to the type cache.         /// </summary>         /// <param name="type">The type.</param>         /// <param name="proxy">The proxy.</param>         public static void AddProxyForType(Type type, Type proxy)         {             lock (syncRoot)             {                 Cache.Add(GetHashCode(type.AssemblyQualifiedName), proxy);             }         }           /// <summary>         /// Tries the type of the get proxy for.         /// </summary>         /// <param name="type">The type.</param>         /// <returns></returns>         public static Type TryGetProxyForType(Type type)         {             lock (syncRoot)             {                 Type proxyType;                 Cache.TryGetValue(GetHashCode(type.AssemblyQualifiedName), out proxyType);                 return proxyType;             }         }           #region Private Methods           private static string GetHashCode(string fullName)         {             SHA1CryptoServiceProvider provider = new SHA1CryptoServiceProvider();             Byte[] buffer = Encoding.UTF8.GetBytes(fullName);             Byte[] hash = provider.ComputeHash(buffer, 0, buffer.Length);             return Convert.ToBase64String(hash);         }           #endregion     } } As you can see, there are two public methods, one for adding to the cache and one for getting from the cache. Hopefully they should be clear enough, the Get is a TryGet as I do not want the dictionary to throw an exception if a proxy doesn’t exist within the cache. Other than that I’ve decided to create a key using the SHA1CryptoServiceProvider, this may change but my initial though is the SHA1 algorithm is pretty fast to put together using the provider and it is also very unlikely to have any hashing collisions. (there are some maths behind how unlikely this is – here’s the wiki if you’re interested http://en.wikipedia.org/wiki/SHA_hash_functions)   Anyway, that’s the end of part 1 – although I haven’t started any of the fun stuff (by fun I mean hairpulling, teeth grating Relfection.Emit style fun), I’ve got the basis of the DynamicProxy in place so all we have to worry about now is creating the types, interceptor classes, method invocation information classes and finally a really nice fluent interface that will abstract all of the hard-core craziness away and leave us with a lightning fast, easy to use AOP framework. Hope you find the series interesting. All of the source code can be viewed and/or downloaded at our codeplex site - http://rapidioc.codeplex.com/ Kind Regards, Sean.

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  • Another sound not working post

    - by Thomas Smart
    Tried all the other "sound not working" posts i think, lost count. purge/reinstall alsa and pulse, reboot, add user to audio group, various lines in the alsa config file such as "options snd-hda-intel model=" then tried different options like generic, auto, basic, default, etc. tried pulseaudio -k && sudo alsa force-reload a few times, with and without rebooting. Hardware: 16gb ram, core I7-4790, Intel Haswell mboard with onboard sound and graphics Multimedia: Audio Adapter: HDA-Intel-HDA Intel HDMI OS: Ubuntu server 14.04 with ubuntu-desktop installed. GUI sound settings lists only the dummy sound card alsamixer -c 0 ¦ Card: HDA Intel HDMI F1: Help ¦ ¦ Chip: Intel Haswell HDMI F2: System information ¦ ¦ View: F3:[Playback] F4: Capture F5: All F6: Select sound card ¦ ¦ Item: S/PDIF ¦ ¦ +--+ ¦ ¦ ¦OO¦ ¦ ¦ +--+ ¦ ¦ < S/PDIF > ¦ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server hdmi:CARD=HDMI,DEV=0 HDA Intel HDMI, HDMI 0 HDMI Audio Output dmix:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Direct sample mixing device dsnoop:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Direct sample snooping device hw:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Direct hardware device without any conversions plughw:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Hardware device with all software conversions cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7d14000 irq 46 cat /proc/asound/devices 1: : sequencer 2: [ 0- 3]: digital audio playback 3: [ 0- 0]: hardware dependent 4: [ 0] : control 33: : timer mplayer -ao alsa:device=hdmi /usr/share/sounds/ubuntu/stereo/system-ready.ogg MPlayer 1.1-4.8 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/ubuntu/stereo/system-ready.ogg. libavformat version 54.20.4 (external) Mismatching header version 54.20.3 libavformat file format detected. [lavf] stream 0: audio (vorbis), -aid 0 Load subtitles in /usr/share/sounds/ubuntu/stereo/ ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders libavcodec version 54.35.0 (external) AUDIO: 44100 Hz, 1 ch, floatle, 80.0 kbit/5.67% (ratio: 10000->176400) Selected audio codec: [ffvorbis] afm: ffmpeg (FFmpeg Vorbis) ========================================================================== [AO_ALSA] alsa-lib: confmisc.c:768:(parse_card) cannot find card '1' [AO_ALSA] alsa-lib: conf.c:4248:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory [AO_ALSA] alsa-lib: confmisc.c:392:(snd_func_concat) error evaluating strings [AO_ALSA] alsa-lib: conf.c:4248:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory [AO_ALSA] alsa-lib: confmisc.c:1251:(snd_func_refer) error evaluating name [AO_ALSA] alsa-lib: conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory [AO_ALSA] alsa-lib: conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory [AO_ALSA] alsa-lib: pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM hdmi [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hdmi' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video Exiting... (End of file) mplayer -ao alsa:device=hw=0.3 /usr/share/sounds/ubuntu/stereo/system-ready.ogg MPlayer 1.1-4.8 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/ubuntu/stereo/system-ready.ogg. libavformat version 54.20.4 (external) Mismatching header version 54.20.3 libavformat file format detected. [lavf] stream 0: audio (vorbis), -aid 0 Load subtitles in /usr/share/sounds/ubuntu/stereo/ ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders libavcodec version 54.35.0 (external) AUDIO: 44100 Hz, 1 ch, floatle, 80.0 kbit/5.67% (ratio: 10000->176400) Selected audio codec: [ffvorbis] afm: ffmpeg (FFmpeg Vorbis) ========================================================================== [AO_ALSA] Format floatle is not supported by hardware, trying default. AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 0.4 (00.4) of 0.8 (00.7) 0.1% Exiting... (End of file) Thank you for your time and help :)

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  • MP3 fingerprint tagger

    - by droberts
    Does anyone know of a tool which will read mp3 audio information directly (not the tag information), generate a fingerprint of that audio information, recommend tags based on the fingerprint and retag your MP3 collection? Last.FM released a console application which did all but retag your collection.

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  • Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

    - by Roy Rico
    I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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  • Burn 24/96 flac files to play on standalone player

    - by takeshin
    I have vinyl record rip in 24/96 flac format. Each track is almost 200 MB big, so the album won't fit on CD. How to burn these files on a DVD to play with the same quality on standalone DVD player? My player supports SACD, DVD Audio and DVD video as well. My OS is Ubuntu Lucid (preferred), but I have also WinXp with Nero installed. BTW, is there any difference between DVD+ and DVD- for audio?

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  • Replace sound in another YouTube video

    - by Tom
    I have received permission from someone to translate the audio in their movies. The problem I am facing is that the video quality is quite poor and the author does not have the original videos any more. How can I replace the audio in the YouTube videos without further degrading the quality of the videos? Thanks, Tom

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  • Any plugins for Skype that support "Soundboard" usage?

    - by Axxmasterr
    I would like to find a program or plugin for Skype that allows you to pipe sound samples in to the outgoing audio stream when you are on a call. Ideally it would have some sort of soundboard functionality so that I could have a group of audio samples at the touch of a button. I'd also prefer something that supports mp3 but wav support will also do.

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  • Debian sound on hdmi instead of jack

    - by Hans de Jong
    I installed debian (gnome) and i can't get my sound working. When i use inxi -A i get the following result: Audio: Card-1: Advanced Micro Devices [AMD] nee ATI Cayman/Antilles HDMI Audio [Radeon HD 6900 Series] driver: snd_hda_intel Card-2: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) driver: snd_hda_intel Sound: Advanced Linux Sound Architecture ver: 1.0.24 My feeling tells me my sound output is on the HDMI instead of my jackplug on my motherboard. How can i change this to my motherboard sound output?

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  • Which connector do I need for a "line level" subwoofer?

    - by Ben Brocka
    I've got a separate pair of speakers and I'm looking at adding a subwoofer (this, specifically). I noticed on the detail page it's inputs are listed as such: Inputs: Speaker level, line level If I'm not mistaken "line level" are the standard 3.5 audio jacks on your motherboard/sound card, right? My motherboard has the standard 6 ports for sound, if I get a subwoofer like this can I simply plug the input into the orange 3.5 jack? My audio software supports up to 7.1 so software-wise, 2.1 wouldn't be a problem.

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  • Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

    - by Roy Rico
    I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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  • Changing default playback device on Windows 8

    - by emartel
    Previously, on Vista and Windows 7, changing the Default Playback device would occur instantly. For example, audio is coming out of my speakers, I right click the Volume Control, click Playback Devices then I select another device and click Set Default. Audio would be transferred immediately. Unfortunately, now, with Windows 8, I need to kill whatever process what outputting sound, and restart it for the change to take effect. Is there something that can be done about it so that changes are taken into account immediately?

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  • WCF <operation>Async methods not generated in proxy interface

    - by Charlie
    I want to use the Asnyc methods rather than the Begin on my WCF service client proxy because I'm updating WPF controls and need to make sure they're being updated from the UI thread. I could use the Dispatcher class to queue items for the UI thread but that's not what I'm asking about.. I've configured the service reference to generate the asynchronous operations, but it only generates the methods in proxy's implementation, not it's interface. The interface only contains syncronous and Begin methods. Why aren't these methods generated in the interface and is there a way to do this, or do I have to create a derived interface to manually add them?

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  • OpenIndiana (illumos): vmxnet3 interface lost on reboot

    - by protomouse
    I want my VMware vmxnet3 interface to be brought up with DHCP on boot. I can manually configure the NIC with: # ifconfig vmxnet3s0 plumb # ipadm create-addr -T dhcp vmxnet3s0/v4dhcp But after creating /etc/dhcp.vmxnet3s0 and rebooting, the interface is down and the logs show: Aug 13 09:34:15 neumann vmxnet3s: [ID 654879 kern.notice] vmxnet3s:0: getcapab(0x200000) -> no Aug 13 09:34:15 neumann vmxnet3s: [ID 715698 kern.notice] vmxnet3s:0: stop() Aug 13 09:34:17 neumann vmxnet3s: [ID 654879 kern.notice] vmxnet3s:0: getcapab(0x200000) -> no Aug 13 09:34:17 neumann vmxnet3s: [ID 920500 kern.notice] vmxnet3s:0: start() Aug 13 09:34:17 neumann vmxnet3s: [ID 778983 kern.notice] vmxnet3s:0: getprop(TxRingSize) -> 256 Aug 13 09:34:17 neumann vmxnet3s: [ID 778983 kern.notice] vmxnet3s:0: getprop(RxRingSize) -> 256 Aug 13 09:34:17 neumann vmxnet3s: [ID 778983 kern.notice] vmxnet3s:0: getprop(RxBufPoolLimit) -> 512 Aug 13 09:34:17 neumann nwamd[491]: [ID 605049 daemon.error] 1: nwamd_set_unset_link_properties: dladm_set_linkprop failed: operation not supported Aug 13 09:34:17 neumann vmxnet3s: [ID 654879 kern.notice] vmxnet3s:0: getcapab(0x20000) -> no Aug 13 09:34:17 neumann nwamd[491]: [ID 751932 daemon.error] 1: nwamd_down_interface: ipadm_delete_addr failed on vmxnet3s0: Object not found Aug 13 09:34:17 neumann nwamd[491]: [ID 819019 daemon.error] 1: nwamd_plumb_unplumb_interface: plumb IPv4 failed for vmxnet3s0: Operation not supported on disabled object Aug 13 09:34:17 neumann nwamd[491]: [ID 160156 daemon.error] 1: nwamd_plumb_unplumb_interface: plumb IPv6 failed for vmxnet3s0: Operation not supported on disabled object Aug 13 09:34:17 neumann nwamd[491]: [ID 771489 daemon.error] 1: add_ip_address: ipadm_create_addr failed on vmxnet3s0: Operation not supported on disabled object Aug 13 09:34:17 neumann nwamd[491]: [ID 405346 daemon.error] 9: start_dhcp: ipadm_create_addr failed for vmxnet3s0: Operation not supported on disabled object I then tried disabling network/physical:nwam in favour of network/physical:default. This works, the interface is brought up but physical:default fails and my network services (e.g. NFS) refuse to start. # ifconfig -a lo0: flags=2001000849<UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL> mtu 8232 index 1 inet 127.0.0.1 netmask ff000000 vmxnet3s0: flags=1004843<UP,BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:1: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:2: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:3: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:4: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:5: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:6: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:7: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 vmxnet3s0:8: flags=1004842<BROADCAST,RUNNING,MULTICAST,DHCP,IPv4> mtu 9000 index 2 inet 192.168.178.248 netmask ffffff00 broadcast 192.168.178.255 lo0: flags=2002000849<UP,LOOPBACK,RUNNING,MULTICAST,IPv6,VIRTUAL> mtu 8252 index 1 inet6 ::1/128 vmxnet3s0: flags=20002000840<RUNNING,MULTICAST,IPv6> mtu 9000 index 2 inet6 ::/0 # cat /var/svc/log/network-physical\:default.log [ Aug 16 09:46:39 Enabled. ] [ Aug 16 09:46:41 Executing start method ("/lib/svc/method/net-physical"). ] [ Aug 16 09:46:41 Timeout override by svc.startd. Using infinite timeout. ] starting DHCP on primary interface vmxnet3s0 ifconfig: vmxnet3s0: DHCP is already running [ Aug 16 09:46:43 Method "start" exited with status 96. ] NFS server not running: # svcs -xv network/nfs/server svc:/network/nfs/server:default (NFS server) State: offline since August 16, 2012 09:46:40 AM UTC Reason: Service svc:/network/physical:default is not running because a method failed. See: http://illumos.org/msg/SMF-8000-GE Path: svc:/network/nfs/server:default svc:/milestone/network:default svc:/network/physical:default Reason: Service svc:/network/physical:nwam is disabled. See: http://illumos.org/msg/SMF-8000-GE Path: svc:/network/nfs/server:default svc:/milestone/network:default svc:/network/physical:nwam Reason: Service svc:/network/nfs/nlockmgr:default is disabled. See: http://illumos.org/msg/SMF-8000-GE Path: svc:/network/nfs/server:default svc:/network/nfs/nlockmgr:default See: man -M /usr/share/man -s 1M nfsd Impact: This service is not running. I'm new to the world of Solaris, so any help solving would be much appreciated. Thanks!

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  • pfSense 2.1 OpenVPN client not using tunnelled interface

    - by Brian M. Hunt
    I'm having some trouble getting OpenVPN working on my pfSense box. The issue is quite strange to me. When I have the OpenVPN turned on, only my router is able to connect to the Internet. From the router I can use ping, links, etc., and connections work exactly as expected - through the VPN, with the IP address assigned by my VPN provider (Proxy.sh, incidentally). However, none of the clients on the local network can connect to the Internet. I get timeouts when using ping or a web browser. I can ping my router, and the IP address of the gateway. When I switch the default gateway from the VPN to my ISP's gateway, all works exactly as expected. Here the routing table (netstat -r) when in VPN mode, and a key for it: IPv4 Destination Gateway Flags Refs Use Mtu Netif Expire 0.0.0.0/1 10.XX.X.53 UGS 0 122 1500 ovpnc1 = default 10.XX.X.53 UGS 0 235 1500 ovpnc1 8.8.8.8 10.XX.X.53 UGHS 0 82 1500 ovpnc1 10.XX.X.1/32 10.11.0.53 UGS 0 0 1500 ovpnc1 10.XX.X.53 link#12 UH 0 0 1500 ovpnc1 10.XX.X.54 link#12 UHS 0 0 16384 lo0 ZZ.XX.XXX.0/20 link#1 U 0 83 1500 re0 ZZ.XX.XXX.XXX link#1 UHS 0 0 16384 lo0 127.0.0.1 link#9 UH 0 12 16384 lo0 128.0.0.0/1 10.11.0.53 UGS 0 123 1500 ovpnc1 192.168.1.0/24 link#11 U 0 1434 1500 ue0 192.168.1.1 link#11 UHS 0 0 16384 lo0 YYY.YYY.YYY.YYY/32 ZZ.XX.XXX.1 UGS 0 249 1500 re0 IP addresses 10.XX.X.53/54 - My DHCP-assigned IP address/pair from the VPN provider ZZ.XX.XXX.XXX - My external IP assigned by my ISP YYY.YYY.YYY.YYY - The external IP assigned by the VPN provider Interfaces ovpnc1 - My VPN client interface re0 - My LAN interface ue0 - My WAN interface This looks essentially what I would expect it to be. The default route is through the VPN provider. The VPN address is routed through the ISP-assigned IP address. I am not sure what would be wrong here. So figuring this was a firewall issue, I basically tried enabling all in/out traffic. This did not seem to remedy the problem. Also figuring it could possibly be some client networking issue, I restarted the clients on the LAN. This did not help. I also ran route flush and reset the routes manually. So I am a bit stumped, and would be very grateful for any thoughts on what the problem might be.

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