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Audio booster Pack 8Bit 8000Khz Mono

Maurice Rozema
Prodigy 140 points
Other Parts Discussed in Thread: CC3200, TLV320AIC3254, TLV320DAC32, TLV320AIC32

Hello,

I have a problem with the CC3200 in combination with the AUDIO booster pack(TLV320AIC3254).

At the moment the bitrate for sound is 512000 (which can't go lower, because the hardware can't handle a lower clockspeed), but how can i sample the sound at a bitrate of 512000 a second and then convert it to 64000.

512000 = (16*16000*2) = (BITS_PER_SAMPLE*SAMPLERATE*STEREO)

64000 = (8*8000*1) = (BITS_PER_SAMPLE*SAMPLERATE*MONO)

I want to start the AudioCodec with 8BIT, 8000 samples a second and in mono sound. But the internal clock can't handle the 64000 bit per clock speed. So how can I generate 16BIT, 16000, Stereo sound in the audio codec and downsample it to 8BIT, 8000, Mono sound in the RAW PCM circular buffer or in the DMA.

Thank you in advance,

Maurice

  • 0
    TI__Guru* 91206 points
    Maurice -
    the TLV320AIC3254 samples between 8kHz and 192kHz for stereo and at 8kHz for mono.
    the connection between the this device and the CC3200 (Boosterpack to LaunchPad) is I2C, which can go up to 400kHz in speed.

    can you point to the file in the SDK example in which you found the snippet above?
  • 0 in reply to Josh Wyatt
    TI__Mastermind 27080 points
    It seems, this querry is posted twice.. Here is the other thread -
    e2e.ti.com/.../539277.

    As recommended in the other thread, for changing the audio sample rate -

    a) Find, out the audio codec (AIC3254) register setting needed for desired audio sample frame rate and program accordingly. I believe, there are offline tools from Audio Codecs team to generate the needed register settings for desired audio sample rate, audio filter selection on codec etc...

    Other option -

    b) If, you are looking for 8KHz sample rate, mono, 'down sample' the 16KHz Audio samples from audio FIFO, by programming the uDMA to skip every other audio FIFO buffer location when, DMAing to application audio buffer..

    Thanks
  • 0 in reply to Josh Wyatt
    Prodigy 140 points

    Hello Josh and Hari,

    I think the TLV320AIC3254 can't sample in 8kHz. What I want is to configure the audio codec with this settings:

    8Bit, 8kHz, Mono, but then the bit clock speed will be 64000 (64000 = (8*8000*1) = (BITS_PER_SAMPLE*SAMPLERATE*MONO)), which seems to be impossible if we look at this document:

    "The TLV320AIC32/33/31/3101/3104/3105/3106/3204/3254 codecs and TLV320DAC32 digital-to-analog converter (DAC) from TI provide a glueless interface to applications with McBSPs. The digital audio interface in these devices is programmable to work with popular audio standard protocols (I2S, DSP, Left-/Right-Justified, and TDM) and 16-, 20-, 24- and 32- bit data widths. Furthermore, Word-Clock (WCLK) and Bit-Clock (BCLK) can be independently configured in either Master or Slave Mode for flexible connectivity to a wide variety of processors. An on-chip PLL enables generation of audio clocks from a variety of system clocks from 512 kHz to 50 MHz."

    Text from: www.ti.com/.../slaa413.pdf

    As of the example in the SDK, I am using "wifi_audio_app", i didn't edit the audiocodec settings so which snippet are you pointing towards?

    and Hari as you stated in this sentence: "offline tools from Audio Codecs team to generate the needed register settings". Where can i find them, I did a search but couldn't find anything.

    -OR-


    The problem isn't in the sound, the quility is great but, the problem lies in the bandwidth over the network. At the moment the bandwidth is 512Kbps. This is way to high for non-stop audio streaming. How can I lower the bandwidth without lowering the audi codec settings. Do i have to compress the raw PCM data (from the circular buffer) to an encoder? so i can lower the bandwidth over the network? (This isn't my cup of Tea, so it is hard to tell you more about it.)

    Thanks in advance,

    Maurice