Real Time Audio Transmission In CELT Using GNU Radio By
International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) International Research Publication House http://www.irphouse.comReal Time Audio Transmission in CELT using GNU Radio By USRP2Author1: Mohammad Mosfiqur RahmanInstructor, Department of Computer TechnologyBarisal Polytechnic Institute, Barisal, BangladeshAuthor2: Dr Miftahur RahmanProfessor, Department of Mathematics and PhysicsNorth South University, Dhaka, BangladeshAbstractThis paper presents the issue of the innovation policy of realtime audio transmission process in CELT codec using GNUradio, a SDR by USRP2. As process technology evolves,processors become more computationally capable pushingthe borderline between software and hardware closer to theantenna. To serve the purposes of innovation LPC isnecessary for speech coding. The basic foundation of speechcoding is to represent the speech signal with the fewestnumber of bits, while maintaining a sufficient level ofquality of the retrieved or synthesized speech withreasonable computational complexity. To achieve highquality speech at a low bit rate, coding algorithms applysophisticated methods to reduce the redundancies, that is, toremove the irrelevant information from the speech signal.The paper presents a comprehensive assessment of theinnovation process that, audio transmission using CELP is agood codec, but for batter performance in real time audiotransmission CELT is the one of the best codec cause of itscodebook and other resources. In this innovation CELT isthe perfect combination with GNU radio for this purposes.read/write from/to in different formats, like binary complexvalues or WAV-files. This Graphical User Interface (GUI)can be used to recreate any model based on the need. TheGRC helps to easily connect the different modules withoutthe need of using the command line interface or directlywriting the python codes.GMSK (Gaussian Minimum Shift Keying) Modulation is amodulation technique that can provide large data rates withsufficient robustness to radio channel impairments.GMSK had also achieved popularity for use in commercialhigh speed broadband wireless systems as the spectrum isutilized more efficiently. GMSK is now being widelyimplemented in high-speed digital communications. GMSKhas been accepted as standard in several wire line andwireless applications. Thus GMSK is the next generationtransport technology for wireless communications. GMSK isa special form of multicarrier modulation technique inwhich the available bandwidth is divided into many narrowsub carriers or sub-channels. This allows many users totransmit in an allocated band in an GMSK system. Each useris allocated several carriers in which to transmit their data.The separation of the sub-carriers is such that there is a verycompact spectral utilization. With GMSK, it is possible tohave no overlapping sub channels in the frequency domain,thus increasing the transmission rate. ( Gina Colangelo) [1]With a SCA (Software Communications Architecture)implementation like GNU radio project has emerged as oneof the most exciting and promising technology. The GNUradio system provides an open source software platformwhich together with low cost hardware called USRP(Universal Software Radio Peripheral) can be used todevelop various software radio applications and implementnew technologies for testing purpose. The Software DefinedRadio (SDR) allows to bring the code as close to the antennaas possible and because of this it becomes more convenientto be used for academic purposes. The code were generatedusing C , Python and XML, which the includes theprocesses involved in formation of the GMSK signal fortransmission in modulation techniques and alsodemodulation techniques for the received signal. Here is atechnique where more than one sub carriers are used totransmit a single data. (Mutsawashe Gahadza , MinseokKim, Jun-Ichi Takada,)[2]Keyboard : GNU Radio, LPC,CELP, CELT, SDR, USRP2,GMSK, Sampling rate,IntroductionMost of the wireless system research uses the simulation asan important tool to validate the system performance. Themotivation of this topics is to build a flexible test bed forevaluating the novel algorithms under wireless transmissionenvironment. The rapid growth in wireless communicationsystems demands a technology that is capable of conveyingdata at high speed and with reliability. The future ofcommunication is wireless, therefore both research andtesting focus on improvement of the techniques of wirelesstransmission. In GNU radio GNU Radio Companion (GRC)is a graphical tool for creating signal flow graphs andgenerating flow-graph source code. It is an open-sourceVisual programming language for signal processing usingthe GNU Radio libraries. The GNU Radio package isprovided with a complete HDTV transmitter and receiver, aspectrum analyzer, an oscilloscope, a multichannel receiverand a wide collection of modulators and demodulators. Theuser interface is called GNU Radio companion or GRC.GNU Radio has several blocks that can generate data or445
International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) International Research Publication House s such as remote musical collaboration even overlong distances. In keeping with the Xiph.Org mission—CELT is also designed to accomplish this without copyrightor patent encumbrance. Only by keeping the formats thatdrive our Internet communication free and unencumberedcan we maximize innovation, collaboration, andinteroperability. . There is also a basic tool for testing theencoder and decoder called68 "testcelt" located in libcelt/:testcelt rate channels frame size bytes per packet input.sw output.sw, where input.sw is a 16-bit (machineendian) audio file sampled at 32000 Hz to 96000 Hz. Theoutput file is already decompressed. For example, for a 44.1kHz mono stream at 64kbit/sec and with 256 sampleframes: testcelt 44100 1 256 46 intput.sw output.sw Since44100/256*46*8 63393.74 bits/sec.All even frame sizesfrom 64 to 512 are currently supported, although power-oftwo sizes are recommended and most CELT development isdone84 using a size of 256. The delay imposed by CELT is1.25x - 1.5x the frame duration depending on the frame sizeand some details of CELT's internal operation. For 256sample frames the delay is 1.5x or 384 samples, so the totalcodec delay in the above example is 8.70ms(1000/(44100/384)).CELT is already ahead of the competition. Its delay:Configurable, 1.3 ms to 24 ms, 8 ms typical and quality (atequivalent rates): Much better than G.722.1C, as good as orbetter than AAC-LD, better than ULD. Its flexibility: 24kbps to 160 kbps, 32 kHz to 96 kHz, configurable delay,low-complexity mode The freedom: Open source (BSD), nopatents and the transform codec (MDCT, like MP3, Vorbis)Explicitly code energy of each band of the signal has coarseshape of sound preserved no matter what and coderemaining details using vector quantization. Also uses pitchprediction with a time offset, CELT is similar to linearprediction used by speech codecs and helps compensate forpoor frequency resolutionCELT is short block transform that only capable ofresolving harmonics if the period is an exact multiple of theframe size. For any other period length, the current windowwill contain a portion of the period offset by some phase.We search the recently decoded signal data for a windowthat covers the same portion of the period with the samephase offset. While the harmonics will still not resolve intodistinct MDCT bins, for periodic inputs the predictor willproduce the same pattern of energy spreading. The pitchpredictor is specified by a period defined in the time domainand a set of gains defined in the frequency domain. Thepitch period is the time offset to the window in the recentsynthesis signal history that best matches the currentencoding window. We estimate the period using thefrequency domain generalized cross-correlation between thezero-padded input window and the last Lp 1024 decodedsamples .Two of the parameter sets transmitted to the decoder areencoded at variable rate: the energy in each band, which isentropy coded, and the pitch period, which is not transmittedif the pitch gains are all zero. To achieve a constant bit-ratewithout a bit reservoir, we must adapt the rate of theinnovation quantization. We first assume that both theencoder and the decoder know how many 8-bit bytes arePC2Figure 1 :Communication model with USRPImplementing the GMSK signal and thus a better signal canbe retrieved which would otherwise had been distorted andultimately lost. Transmitters of this type use GMSKmodulation and digital encoding to guarantee protection oftransmitted data. Only special receiver, equipped withrelevant decoder, can receive signals from such transmitters.Any other receiver provides ―white noise‖ reception only.GMSK is a simple yet effective approach to digitalmodulation for wireless data transmission. GMSK has beenadopted by many wireless data communication protocols.Key advantages include spectral efficiency, low phasedistortion and coherence of the signal, it also improves noiseimmunity when demodulating.MethodsIn most modern paper in the field of audio transmission themethodological concepts competitively as a set of rules andpractice followed by codec. CELT (Constrained EnergyLapped Transform) is an open, royalty-free audiocompression format and a free software codec for use inlow-latency audio communication. It is a lossy codec, utputis split in bands approximating the critical bands; . (Dr.Jean-Marc Valin,Gregory Maxwell, and Dr. Timothy B.Terriberry)[3]Sampling rates from 32 kHz to 48 kHz and above can be usein CELT, adaptive bit-rate from 32 kbit/s to 128 kbit/s perchannel and above. It uses ultra-low algorithmic delay (aslow as 2 ms; scalable, typically from 3 to 9 ms).One of the very low delay audio codec CELT designed forhigh-quality communications. Traditional full-bandwidthcodecs such as Vorbis and AAC can offer high quality butthey require codec delays of hundreds of milliseconds,which makes them unsuitable for real-time interactiveapplications like tele-conferencing. Speech targeted codecs,such as Speex or G.722, have lower 20-40ms delays buttheir speech focus and limited sampling rates restricts theirquality, especially for music. Additionally, the othermandatory components of a full network audio system—audio interfaces, routers, jitter buffers— each add their owndelay. For lower speed networks the time it takes to serializea packet onto the network cable takes considerable time,and over the long distances the speed of light imposes asignificant delay. In teleconferencing— it is important tokeep delay low so that the participants can communicatefluidly without talking on top of each other and so that theirown voices don't return after a round trip as an annoyingecho. indeed a challenging area in audio codec design,because as a codec is forced to work on the smaller chunksof audio required for low delay it has access to lessredundancy and less perceptual information which it can useto reduce the size of the transmitted audio. CELT isdesigned to bridge the gap between "music" and "speech"codecs, permitting new very high quality teleconferencingapplications, and to go further, permitting latencies muchlower than speech codecs normally provide to enable446
International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) International Research Publication House http://www.irphouse.comMultiply Constant, Delay, Audio Sink are used. Here samplerate of the UHD: UHD Source is 195.312 KHz and samplerate of audio sink is 48KHz.The output of this transmission& receiver’s quality is still poor in quality. Still noise is inthe output signal, delay, echo is in the signal, so next choicewas GMSK.Using Gaussian Modulation Shift Keying the transmissionside of the GRC blocks are TCP source, Packet Encoder,GMSK Mod, UHD:UHD Sink . There the sample rate ofpacket encoder is 512KHz and the sample rate ofUHD:UHD Sink is 195.312KHz . Address of TCP Source is127.0.0.1.In receiving side the GRC blocks UHD:UHDSource, GMSK Demod , WX GUI Scope Sink, PacketDecoder, TCP Sink are used. Here sample rate of the UHD:UHD Source is 195.312 KHz and address of TCP sink is127.0.0.1.The output of this transmission & receiver’squality is not so good quality. Still noise is in the outputsignal, delay, echo are lightly in the signal. These arehappened for miss match of GNU radio internal code whichautomatic generated when circuit is design in the GRC andour code . So there is a opportunity to make it better byusing code directly.used to encode the frame. This number is either agreed onwhen establishing the communication or obtained during thecommunication, e.g. the decoder knows the size of any UDPdatagram it receives. Given that, both the encoder and thedecoder can implement the same mechanism to determinethe innovation bit allocation. This mechanism is basedsolely on the number of bits remaining after encoding theenergy and pitch parameters. A static table determines thebit-allocation in each band given only the number of bitsavailable for quantizing the innovation. The correspondencebetween the number of bits in a band and the number ofpulses is given by the [6]. For a given number of innovationbits, the distribution across the bands is constant in time.This is equivalent to using a psychoacoustic masking.Each band's share of available bits is fixed, specially CELTtransmits no side information for allocation and it equivalentto modeling within band masking. The signal-to-mask ratiofor each band is roughly constant, so ignores inter-bandmasking and tone vs. noise effects.In this communication one pc with Linux Ubuntu haveproper setup of GNU Radio in tx side and rx side both.These pc are connected with USRP2 N210 . Thecommunication media was wireless communication with 2.4GHz frequency.(AdvanceLinux A (AdvanceCELTLinux SoundArchitecture)PythonDecoderCode(Rx)USRP(GNU Radio)USRP(GNU Radio)Figure2: Basic Architecture of CELT implemented in Linux platformIn present it is become truth that the USRP-2 (DaughterBoard XCVR 2450) is capable to transmit & receive realtime audio signal. It becomes true when in GRC the NBFMmodulation is used in tx & rx end.Using narrow band frequency modulation the transmissionside of the GRC blocks Audio source, Rational Resampler,NBFM transmit, UHD:UHD Sink are used. There thesample rate of audio source is 48KHz and the sample rate ofUHD:UHD Sink is 195.312KHz. And in receiving side theGRC blocks UHD:UHD Source, NBFM Receiver, WX GUIScope Sink, Rational Resampler, Multiply Constant, Delay,Audio Sink are used. Here sample rate of the UHD: UHDSource is 195.312 KHz and sample rate of audio sink is48KHz.The output of this transmission & receiver’s qualitywas so poor in quality. Very much noise is in the outputsignal. Delay, echo is in the signal, so next choice wasWBFM.Using wide band frequency modulation the transmissionside of the GRC blocks are Audio source, WX GUI ScopeSink ,Rational Resampler, WBFM transmit, UHD:UHDSink. There the sample rate of audio source is 48KHz andthe sample rate of UHD:UHD Sink is 195.312KHz.Inreceiving side the GRC blocks UHD:UHD Source, WBFMReceiver, WX GUI Scope Sink, Rational Resampler,Figure3 : Implemented Structure of GNU RadioFor analysis of CELT performance should have some tasksfor1.2.3.4.5.Measure the highest energy with respect tonormalized energyMeasure the fft of .wav file we get in output.Divide others quality respect to the highest quality.Find out the best quality energy.Energy change at the respect of frequency change.There are two lossy audio codecs that are being usedpresently. These are classified in two broad classes.:1. Low delay (15-30 ms) speech codecs that includeG.72x,GSM,AMR,Speex with low sampling rate of 8 KHzto 16 KHz with limited fidelity. These codecs do not supportmusic due to low sampling rate. Low delay is a criticalfactor for live interaction due to low collision rate duringconversation and reduced echo cancellation. Low delaycodecs are suitable for live music synchronization thatrequires delay of less than 25ms.[5]447
International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) International Research Publication House http://www.irphouse.com2. General purpose codec’s that include MP3, AAC, Vorbiswith high delay ( 100ms) and high sampling rate of 44.1KHz and higher. These codec’s support CD quality musicwith higher fidelity.Therefore in summarization the two above mentioned codeccategories, one may observe the following advantages anddisadvantages:i. G.722.1C (ITU-T) with 40 ms delay and upto 32 KHzsampling frequencyii. AAC-LD (MPEG) with 20-50ms delay and samplingfrequency up to 48KHz.iii. ULD (Franhofer) with delay less than 10ms andsampling frequency up to 48 KHz.iv. On the other hand CELT is an open source codec with alot of potential to be competitive with respect to the existingcodec. It has configurable delay in the range of 1.3 ms to 24ms with much better quality than G.722.1C, AAC-LD, andULD. The data rate range from 24kbps to 160kbps andhigher.Figure 6: Energy variation with sampling rate 32 KHzResultIn this part the transmitted signal is being transformed usingFFT to get the magnitude. Though FFT is in different pointso magnitude will act as an energy. Here the audiblefrequency have much more energy as can be seen from Fig 4to Fig 8The sampling rate of the original signal is 44.KHz,then it is verified with 8 KHz, 16 KHz, 32 KHz and 41 KHz.The result of energy versus frequency as functions ofsampling frequencies of 8 KHz,16 KHz, 32 KHz,41 KHzand 44.1 KHz are shown in Fig 4 to Fig 8 respectively.when we find out these energy variations with differentsampling rate then we find out the normalized energy overthe sampling rates. The normalized energy variation oversampling rate in GMSK is shown in Figure 9Figure 7: Energy variation with sampling rate 41KHzFigure 8: Energy variation with sampling rate 44.1KHzNormalized Energy versus Sampling Rate1.2Normalized Energy1Figure 4: Energy variation with sampling rate 8 KHz0.80.6Normalized Energy0.40.20816324144.1Sampling Rate (kHz)Figure 9 Normalized energy versus sampling rateIn GMSK data always transmit from one side to another.When the modulation period come then there will somesmall chunk. The number of byte in every chunk is packetsize. When a packet size increases then GMSK packetencoding delay will also increases. When packet size isFigure 5: Energy variation with sampling rate 16 KHz448
International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) International Research Publication House http://www.irphouse.comsmall then in GMSK there is a calculation in between headerand others tasks of modulation for every part. Sothroughput become decrease. Here it has been taken adifferent number of packet size like : 26bytes, 28 bytes, 30bytes, 32 bytes, 34 bytes, 36 bytes 38 bytes,40 bytes, 42bytes, 44 bytes, 46 bytes, 48 bytes, 50 bytes. Then wetransmit the data and receive it from receiver. The receiverpc shows the maximum throughput when it received dataafter 5 second average in each calculation showed in Table 1Table 2: Throughput of audio Tx & Rx chain of varioussample ratePacketsizeconstantSampleRateRequiredBit rateAchievedBitratefor 5 .966.64YesNo(Bytes)Table 1: GMSK maximum throughputMaximumthroughput inkilobitMaximumthroughput 234.1542.Bit Rate versus Sampling Rate in Audio Throughput of Tx &Rx987Bit Rate (kBps)Packet size in Bytes65Required Bit Rate4Achived Bit Rate32103638404141.542444648Sampling Rate (kHz)5059.617.45Figure 11: Bit rate vs sampling rate in audio throughput of tx & rxAfter getting the maximum GMSK throughput in thisvariation of different packet size, we make a chart. In Fig 10there we get that the upward position of GMSK maximumthroughput when packet size is 44 bytes.In Table 3 and in Fig 12 sampling rate is 40 kHz with 256sample per frame and different size of packet achievedbitrates is nearly similar as well as the required bitrates. Foraudio throughput sample per frame 256 is a optimum pointof compression and delay. From this table we found herethat how many data encoded by 256 sample . If the numberof byte is increases then quality become well, but we have totransmit more data in network transmission. Here it needsuch fixed byte per frame which can keep sufficient level ofquality and performance.GMSK Maximum Throughput9Throughput (kBps)8765Maximum Throughput43Table 3: Throughput of audio Tx & Rx chain of variouspacket size:210262830323436384042444648SamplingRate kHz50Packet Size (Bytes)SamplesperframePacketsize(Byte)Requiredbit rate(kBps)Achivedbit ratefor 5 Figure 10: GMSK Maximum ThroughputIn Table 2 and also in Fig 11 Bit rate (kbps) versussampling rate (kHz) is shown with constant packet size( 42bytes) and sample rate is changing to find out the necessarythroughput and measure the specific sample rate getting inreal time audio transmission in GNU Radio. In this casedelay is not available if there is under flow, because whenrequired bit rate is less then achieved bit rate then underflowhappen. Achieve bit rate is not flat in a certain point becausethe maximum achieved bitrates will be in a maximumsampling rate and it is the maximum speed for transmissionusing GNU radio for a particular packet size40449256
International Journal of Engineering Research and Technology. ISSN 0974-3154 Volume 10, Number 1 (2017) International Research Publication House http://www.irphouse.comalgorithm (requires strong DSP knowledge) or buildingapplications using CELT. Our feedback can help define thefuture direction the codec will take. It applies some of theCELP principles, but does everything in the frequencydomain, which removes some of the limitations of CELP.CELT is suitable for both speech and music [4]There are two program languages used in GNU Radio, C and Python which play different roles in the whole system.All the signal processing and performance-critical blocks arewritten in C . Python is used to create a network or graphand glue these blocks together.Bit rate versus Packet Size in Audio Throughput of Tx & Rx98Bit Rate (kBps)765Required Bit Rate4Achive Bit Rate32103236384042444648Packet Size (Byte)ConclusionThe real time audio transmission using CELT was donesuccessfully using GNU radio system with its requiredhardware and software components. The result shows us thesignificant performance of the model used to transmit audiotransmission in CELT. From the results and analysis it isquite clear that the GNU Radio is usable for real life audiotransmission model since the results showed that the originalsignal could be retrieved almost without noises. The modelwas tested in 2.4 GHz with CELT. CELT brings CD-qualitysound to VoIP-style low-delay applications and better thanMP3 and 10 ms delay.Future work in this area might look into the application ofGNU Radio in consumer facing applications. Exploringextremely low latency transport-layer-adaptive synchronizedaudio-video transmission might also be of interest. Aspectsof CELT that can be improved include dynamic rateallocation, stereo coupling and pitch prediction. Thetransport layer software can also be tested at other centerfrequencies such as 5GHz. Some emerging uses of softwaredefined radio are 4G LTE, WiMax, WiFi, Digital TV,HDTV, mobile TV etc. The application of low latencycodecs such as CELT in these areas could lead togroundbreaking results. The combined application ofexceptional codecs such as CELT and adaptivecommunication systems such as GNU Radio canrevolutionize the communications sector.Figure 12: Bit rate vs packet size in audio throughput of tx & rxIn Table 4 the limit for sample per packet of CELT is 64 to512. If the number of sample per packet increases to 512then the encoding delay will increase. Here encoding delaymeans the time to encode a packet means a sound filetransmit in the CELT and come backTable 4: Constant sample rate vs various frame sizeSample perFrame 8Good256Best256In Table 5 the fixed sampling rate in different frame sizewith different packet size the delay is not available whenthere is underflow. when packet size increasing the delay iscoming and underflow has decreasing. If delay is presentthen echo also present and if delay is absent then echo isalso absent.Table 5: Constant sample rate in different frame size invarious packet esI.Gina Colangelo ―Introduction to GMSK:Gaussian Filtered Minimum Shift Keying‖(EE194 – SDR)II.Mutsawashe Gahadza , Minseok Kim, Jun-IchiTakada, ―Implementation of a Channel Sounderusing GNU Radio Opensource SDR Platform‖,Graduate School of Engineering, Tokyo Institute ofTechnology, 2009.III.Dr. Jean-Marc Valin,Gregory Maxwell, and Dr.Timothy B. Terriberry ―CELT: A Low-latency,High-quality Audio Codec “, 2010IV.Jean-Marc Valin, Member, IEEE, Timothy B.Terriberry, Christopher Montgomery, GregoryMaxwell ― A High-Quality Speech and AudioCodec With Less than 10 ms Delay” 2010V.Jean-Marc Valin, Timothy B. Terriberry, GregoryMaxwell―A FULL-BANDWIDTH AUDIOCODECWITH LOW COMPLEXITY ANDVERY LOW DELAY” 2010VI.www.gnuradio.org, February, 2011DiscussionCELT exploits the fact that the ear is mainly sensitive to theamount of energy in each critical band. The MDCTspectrum is thus divided into 20 bands of roughly onecritical band each, although the lower frequency bands arewider due to the low MDCT resolution. We refer to thesebands as the energy bandsCELT is still in an early state of development. At this point,two ways of getting involved are: helping design the450
the GNU Radio libraries. The GNU Radio package is provided with a complete HDTV transmitter and receiver, a spectrum analyzer, an oscilloscope, a multichannel receiver . and a wide collection of modulators and demodulators. The user interface is called GNU Radio companion or GRC. GNU Ra
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Connect to the audio connector of a camera if the camera supports audio recording. Note: To make a video backup with audio, make sure the camera which supports the audio function is connected to the video-in channel and audio-in channel. For example, the audio data from audio CH1 will be recorded with the video data from video CH1. 3) AUDIO OUT
1.1 Hard Real Time vs. Soft Real Time Hard real time systems and soft real time systems are both used in industry for different tasks [15]. The primary difference between hard real time systems and soft real time systems is that their consequences of missing a deadline dif-fer from each other. For instance, performance (e.g. stability) of a hard real time system such as an avionic control .
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To configure 4/5.1/7.1-channel audio, you have to retask either the Line in or Mic in jack to be Side speaker out through the audio driver. 1-1 Configuring 2/4/5.1/7.1-Channel Audio The motherboard provides five audio jacks on the back panel which support 2/4/5.1/7.1-channel (Note) audio. The picture to the right shows the default audio jack
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further prior to the receiver "brick wall"audio filter Band Pass filter Band Pass filter Tayloe Quadrature Detector Tayloe Quadrature Detector "I"Audio Preamp "I"Audio Preamp RC Active Audio Filter RC Active Audio Filter VFOVFO 300 -500 KHz BW 1.5 KHz BW 1.5 KHz BW 500 Hz BW (CW) "Q"Audio Preamp "Q"Audio Preamp 90 Degree .
HTML5 provides a standard for playing audio files. Audio on the Web Before HTML5, there was no standard for playing audio files on a web page. Before HTML5, audio files could only be played with a plug-in (like flash). The HTML5 audio element specifies a standard way to embed audio in a web page. HTML5 Audio Tags Tag Description
