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The DMP 128 Plus C V model features a 52x48 mix matrix, which includes 4 additional aux outputs to allow for routing of VoIP channels. The DMP 128 Plus and DMP 128 Plus C models feature an extensive 52x44 mix matrix allowing all 12 analog inputs, 8 aux inputs, 16 expansion bus inputs and 16 virtual bus returns to be discretely routed to any or all of the 8 analog outputs, 4 aux outputs, 16 expansion bus outputs and the 16 virtual bus sends. Extensive Mix Matrix and Flexible Routing Within the DMP 128 Plus Designers can quickly get a snapshot view of the entire audio system, including processing blocks and routing assignments, all at once. The flexible on‑screen layout offers fast access to all of the digital audio signal processing tools. The power of ProDSP is easily harnessed with the DSP Configurator Software. ProDSP is loaded with powerful, easy‑to‑configure tools to control level, adaptive gain, automixing, dynamics, filters, delay, ducking, loudness, feedback suppression, and AEC. ProDSP also utilizes studio grade 24‑bit audio converters with 48 kHz sampling to maintain audio signal transparency. ProDSPĮxtron’s exclusive ProDSP is engineered from the ground up using a powerful 64‑bit floating point DSP engine to provide very wide dynamic range and reduce the potential for clipping.
Aec acoustic echo cancellation series#
DMP 128 Plus Series processors can be used anywhere from a credenza-based system to a large multi-rack system, and even in a large, complex, decentralized multi-building system. The AT models provide Dante® audio networking technology with connectivity for up to 48 digital audio inputs and 24 digital audio outputs. The V models also include up to eight independent channels of VoIP, supporting Session Initiation Protocol - SIP 2.0. The DMP 128 Plus Series is equipped with 12 analog mic/line inputs, eight analog outputs, up to four channels of digital audio input and output via USB, up to eight audio file players, an ACP bus for audio control panels, and new configurable macros. The frame size in that case must be a multiple of the partition (block) length, thereby greatly reducing the latency for long impulse responses.The DMP 128 Plus Series is the next generation of Digital Matrix Processors featuring Extron ProDSP™ 64-bit floating point technology. Latency may be reduced by using partitioned FDAF, which partitions the filter impulse response into shorter segments, applies FDAF to each segment, and then combines the intermediate results. This can be unacceptable for many real-world applications. Traditional FDAF is numerically more efficient than time-domain adaptive filtering for long impulse responses, but it imposes high latency, because the input frame size must be a multiple of the specified filter length. Without such detection schemes, the performance of the system with the larger step size is not as good as the former, as can be seen from the ERLE plots. To deal with this performance difficulty, acoustic echo cancelers include a detection scheme to tell when near-end speech is present and lower the step size value over these periods. With a larger step size, the ERLE performance is not as good due to the misadjustment introduced by the near-end speech. % Plot near-end, far-end, microphone, AEC output and ERLEĪECScope2(nearSpeech, micSignal, e, erledB) % Send the speech samples to the output audio deviceĮrle = diffAverager((e-nearSpeech).^2)./ farEchoAverager(farSpeechEcho.^2) % Stream processing loop - adaptive filter step size = 0.04 while(~isDone(nearSpeechSrc)) From the plot, observe that you achieved about a 35 dB ERLE at the end of the convergence period.ĪECScope2.Title = 'Output of Acoustic Echo Canceller mu=0.04' ĪECScope2.Title = 'Echo Return Loss Enhancement mu=0.04' Since you have access to both the near-end and far-end speech signals, you can compute the echo return loss enhancement (ERLE), which is a smoothed measure of the amount (in dB) that the echo has been attenuated. % Far-end speech signal echoed by the roomįarSpeechEchoSrc.SamplesPerFrame = frameSize
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NearSpeechSrc.SamplesPerFrame = frameSize įarSpeechSrc.SamplesPerFrame = frameSize 'BufferLength', length(x)) ĪECScope1.Title = 'Near-End Speech Signal' ĪECScope1.Title = 'Output of Acoustic Echo Canceller mu=0.025' ĪECScope1.Title = 'Echo Return Loss Enhancement mu=0.025' 'TimeSpan', 35, 'TimeSpanOverrunAction', 'Scroll'. 'LayoutDimensions',, 'TimeSpanSource', 'Property'. 'Method', 'Unconstrained FDAF') ĪECScope1 = timescope(4, fs. % Construct the Frequency-Domain Adaptive FilterĮchoCanceller = dsp.FrequencyDomainAdaptiveFilter( 'Length', 2048.