(If there is a package you would like to see featured here, go to the userinfo page and follow the directions there to submit your entry.) More information on this package can be found on the Debian web site. Some programs require the "-timing" option, see the man page for details. Thus vsound provides a virtual audio loopback cable.Īll this is a long way of saying vsound lets you record the audio output of any program compiled to use OSS, i.e. ) and room equalization over advanced musical effect processors to car audio signal processing. The applications range from active loudspeaker concepts (digital crossovers, bass enhancement. This is precisely what vsound does, using a trick called LD_PRELOAD. The freeDSP is an open-source digital signal processor family for the do-it-yourself community. By redirecting /dev/dsp away from the sound card one can record the byte stream generated by any program. The OSS part of the Linux kernel provides the special file /dev/dsp to let programs access the sound card sounds are generated by writing to /dev/dsp. Want your program to talk to a terminal? Just have it read/write from a special file. For real-time application, the expected delay through the resampler would be 7.9 ms. One of the deep abstractions underlying UNIX and the operating systems it influenced, including GNU/Linux, is that everything should be a file. To do this as shown where I use a requirement of 20 KHz audio bandwidth and 80 dB resampling image rejection, I estimate that 171 taps would be needed for FIR1, 95 taps for FIR2 and 25 taps for FIR3 (as linear phase filters so one multiplier for every 2 taps). It uses sox to convert and save the raw data into the desired file format and can help to convert real audio files to some other non-proprietary format. dataSize = sizeof(rx_pong) // was s_Buffer, etc.This program allows you to record the output of any standard OSS program (one that uses /dev/dsp for sound) without having to modify or recompile the program. dataSize = sizeof(tx_pong) // was s_Buffer, etc. dataSize = sizeof(rx_ping) // was s_Buffer, etc.
dataSize = sizeof(tx_ping) // was s_Buffer, etc. Table_indexR = (table_indexR + 2) % N_period // stride by 2 2 kHz I have moved forward with a ping-pong approach using I2S interrupts and placing the DSP workload (transfer and processing of input samples to output samples) in the while ,ĬLOCK_SetClkDiv( kCLOCK_DivPll0Clk, 0U, true) ĬLOCK_SetClkDiv( kCLOCK_DivPll0Clk, 1U, false)
#Real time dsp – audio loopback update
Do I or can I change the configuration of the callbacks to the Tx waits until the Rx is done with sample processing and can now update the Tx buffer? Out their app notes that I can read?Īlso, I am using MCUXpresso IDE v11.1.1 When I use a logic analyzer to monitor entry to the two callbacks I a lot of "thrashing" between the two callbacks. I would ultimately like to be using the PowerQuad and related CMSIS-DSP functions.
#Real time dsp – audio loopback code
Where should be I redoing the DSP math? What changes to to I2S code can/should be made to maximize the processing time available to the DSP math operations. I2S_RxTransferNonBlocking(base, handle, *transfer) Static void RxCallback( I2S_Type *base, i2s_handle_t *handle, status_t completionStatus, void *userData) I2S_TxTransferNonBlocking(base, handle, *transfer) I2s_transfer_t *transfer = ( i2s_transfer_t *)userData * Enqueue the same original s_Buffer all over again */ Static void TxCallback( I2S_Type *base, i2s_handle_t *handle, status_t completionStatus, void *userData)
The key code as I see it are the two callbacks: I want low latency and the ability to perform DSP math on the signal samples arriving at the ADC side of the wm8904 codec and then send the modified signal samples to the DAC side of the wm8904 codec. My starting point has been the interrupt version example. I currently experimenting with the LPC55S69 i2s examples i2s_interrupt_record_playback and i2s_dma_record_playback examples.