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  • No sound through DisplayPort

    - by Chris Koknat
    I'm trying to connect my Elitebook 8440p laptop to my Samsung HDTV. The laptop does not have a HDMI connection, but it does have DisplayPort. I bought a DisplayPort-to-HDMI adapter here http://www.amazon.com/gp/product/B002CSRFD8/ref=oss_product, and connected it with a 3' HDMI cable. The video shows up fine, but there is no audio. DisplayPort, HDMI, and the adapter all support audio. I contacted HP tech support, who told me to update my sound drivers. I installed the driver and rebooted. Supposedly, I should see a "HD Audio" tab. No luck, even after installing the driver again and rebooting. HP closed the case. I'm using XP Pro.

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  • Speakers will not work after I use USB Headset

    - by Josh K
    I am trying to configure my SteelSeries Siberia V2 Frost USB Headset to work with my 2.0 speakers using a jack. My goal is to find a easy way (no restart) to switch playback from my headset and my speakers and vice-versa. If i plug my headset in and make it the default device then restart my application/web page then the sounds works out of headset. If I switch the default to my speakers and restart apps/web pages then sound does not play. I know my speakers are on because if I configure them through windows and test, the sounds play, and sounds also play when I test it through my audio manager. Even if I unplug my headset, I still cannot get sound out of my speakers unless I restart My audio manager is RealTek HD Audio Manager, Windows 7 x64. I have tried the speaker back, usb front. speaker front, usb in front port. I have not tried speaker back, usb back.

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  • Mini-jack problem with Sony Vaio (running XP)

    - by qftme
    I have a five year old Sony Vaio laptop (vgn-fw31m) that has had impact damage to the audio-output mini-jack for about the last year or so. In a recent discussion with my brother, we wondered whether it would be possible to write a program that would enable windows to use the microphone mini-jack input as the audio-output? As I currently use this laptop for work I am not keen to risk pulling it apart in order to replace the components comprising the audio-out. I therefore 'hope' that a programming solution exists. I would really appreciate any advice on this and eagerly await your response. Kind regards, qftme :)

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  • Recording Interfaces for OS X that are supported/work well?

    - by Troggy
    For os x, I would like to know what other audio production/music recording interface type products people have found to work well with os x? I do not want to know about stuff that only works. I want to know about solid products that work well and are supported well by the company when issues arise. I for example have a M-Audio Firewire Solo recording interface. I have found M-Audio to be a company with great mac support for their products and they integrate well with os x features and apple software. Clarification: I am wondering about the recording interfaces themselves, as in the hardware, that are compatible with os x and supported/work/integrate well.

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  • pgpool2+streaming replication failover on only 2 servers?

    - by aneez
    I am trying to configure pgpool2 and postgresql 9.1 to handle failover. I currently have streaming replication running, and are using pgpool2 for read-only load balancing. I have 2 servers in my setup, both running postgresql - 1 master and 1 slave. The master is also running pgpool2. My question is how do I configure this setup to handle failover? Specifically in the case that the master crashes, and the slave has to take over and run pgpool2 as well. Most documentation and examples I have been able to find assumes that pgpool2 is running on a separate server and thus "never" crashes. I may or may not be attacking the problem using the wrong tools. In my production setup I have a total of 3 identical servers all in independent locations. The main goal of the setup is to achieve a high uptime. Thus failover should be automatic, and bringing a failed node back up should cause only minimal downtime. I want all 3 nodes to be as close to identical as possible, and be able to run with just 1 or 2 nodes available. And if possible I want to use load balancing to improve performance. If anyone can help me gain some insight into how to do this using my current setup or suggest a different/better setup. Thank you!

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  • How do I force Windows to play sound through the speakers only when a USB headset isn't connected?

    - by Phoexo
    I'm using a speaker set connected through the green audio jack and a headset which I connect through USB. My problem is that every time I connect/disconnect my headset, I have to go through a lot of settings/restart some programs to make the sound go through the speakers again. What I want is to have audio play through the headset when it's connected, but if I disconnect the headset, I want the audio to automatically play through the speakers. For example, if I connect/disconnect the headset while listening to music, I have to restart the application to make the music play through the correct speaker/headset, and it shouldn't be that inconvenient. (I found this somewhat relevant topic, but the problem is that it doesn't really give an answer. (Also, it is 2 years old.))

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  • Audio comes out of both headphone and speaker at the same time.. Ubuntu 12.04LTS [closed]

    - by pst007x
    I have the same issue on an Aspire. Ubuntu 12.04LTS 64bit realtek audio sound chip onboard If I plug in a headset, audio does not switch from internal speaker to headset, instead plays out of both at the same time. I have looked at the alsamixer setting, all on. I installed gnome-alsamixer, and I noticed headphone was ticked, if I untick the main audio mutes, and the headphone no longer works. Headset only works with internal speaker. Audio works fine on my other desktop and laptop running this release 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) salvatore@salvatore-Aspire-7730:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. salvatore@salvatore-Aspire-7730:~$ head -n 1 /proc/asound/card*/codec#* ==> /proc/asound/card0/codec#0 <== Codec: Realtek ALC888 ==> /proc/asound/card0/codec#1 <== Codec: LSI ID 1040 ==> /proc/asound/card0/codec#2 <== Codec: Intel Cantiga HDMI salvatore@salvatore-Aspire-7730:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 salvatore@salvatore-Aspire-7730:~$ uname -a Linux salvatore-Aspire-7730 3.2.0-23-generic #36-Ubuntu SMP Tue Apr 10 20:39:51 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux salvatore@salvatore-Aspire-7730:~$ The alsa-base.conf does not exist Tried this: sudo apt-get remove --purge alsa-base sudo apt-get remove --purge pulseaudio sudo apt-get install alsa-base sudo apt-get install pulseaudio sudo alsa force-reload Then: sudo apt-get purge pulseaudio gstreamer0.10-pulseaudio sudo apt-get install pulseaudio gstreamer0.10-pulseaudio indicator-sound Tred this. sudo gedit Then open terminal: sudo /etc/modprobe.d/alsa-base.conf At the end of the file add a new line: options snd-hda-intel model=generic Save and then reboot But alsa-base.conf does not exist

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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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  • 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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  • 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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  • 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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  • Streaming files from EventMachine handler?

    - by Noah
    I am creating a streaming eventmachine server. I'm concerned about avoiding blocking IO or doing anything else to muck up the event loop. From what I've read, ruby's non-blocking IO can be used to stream files in a non-blocking way, or I can call next_tick, but I'm a little unclear about which of these approaches is preferable. Part of the problem is that I have not found a good explanation of non-blocking IO library functions in ruby. Short version: Assuming a long-lived network IO operation, several wall clock minutes of streaming per file, transfer, what is the best way to do this in eventmachine without gumming up the event loop? while 1 do file.read do |bytes| @conn.send_data bytes end end I understand that the above code will block and I'm wondering what to put in its place. Also, I cannot use the FileStreamer class that is part of eventmachine as is, because I need to manipulate the data after it's read but before it's sent. Thanks, Noah

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  • monitoring streaming server and display throughput

    - by I__
    Scenario: laptop (running RHEL 5.3 / 5.4) with Wi-Fi allowing incoming connections (the laptop is the DHCP server and default gateway of any device that connects to it). The laptop has a streaming server installed (my app). I need to program an app that could monitor this link (device / streaming server) and display the throughput. More importantly, I need this app to be able to throttle the throughput. Think WANem but as an app, or netlimiter but (way) simpler and for RHEL. If you need clarifications, let me know. is there a library that could help me? i've done mostly windows business applications programming, and i have no clue about this stuff. please help me to get started!

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  • Netflix streaming stops loading at 98% on Revo 3700

    - by Martin Harris
    I'm trying to stream Netflix on an Acer Revo 3700 running Windows 7 Home Premium, but it hangs on the loading screen at 98% (after it has formatted the player to the right aspect ratio and added the controls, but before the video starts) with no error messages or failures. I have two other machines on the same network, one running Windows 7 Home Premium and another running XP, which both stream faultlessly. Things I have tried: Both a wired and wireless connection to the router Upgrading the video and audio drivers IE, Chrome and Firefox Boxee software Connecting with a VGA cable instead of HDMI (in case it is a HDCP thing) Uninstalling and reinstalling Silverlight. Getting someway into loading a HD movie and turning "Allow HD" off Does anyone know what Netflix is doing at the 98% load mark? Are there any log files? Anything else worth trying? Full disclosure: I'm using Netflix from the UK through a US based VPN. I've tried multiple VPNs and the problem is exactly the same, also the other machines on the same network through the same VPN work fine so I don't think this is the issue, but it might be a factor. The region check happens at around 7% and I get past that.

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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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  • How can I use Android as a remote control for streaming?

    - by michael
    I would like to know if I can use Android to remote control streaming on my laptop? I would like to use my laptop as my streaming server and use my HDTV to view the stream. And I need some way to remote control my streaming server. I have read about http://maketecheasier.com/install-vlc-shares-in-Ubuntu-and-stream-videos-to-Android/2011/02/25 and http://code.google.com/p/android-vlc-remote/ but those are streaming to Android phone itself. I am just need something to remote control streaming to my TV. Is that possible?

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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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