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The first part ...YES but second part NO... we don't know the Audio program, the Audio pipeline used, the elements used and how they are falling and where in the Audio chain of the DSP program they are placed...unless we know that part, we may not be able to change the DSP program, but the first part, yes, there are 4 I2S inputs in AKM7735, and one out of these 4 is used to get I2S PCM data from Bluetooth module, so we may have to solder that I2S line with your new Bluetooth module ...also we can take out the I2S Clock signal lines of the AKM7735 and from the Oscilloscope we can find out the Sampling frequency and Bit clock frequency and from that find out the I2S format ...so no problem with that...only problem we have to solder out around 4-5 pins to be able to do what you are asking for...

Thank you sir. I wish to do this project. Request you to publish the schematics and program.

2 days if used continuously the Grey colored battery which powers the STM32F103 & 7-segments...the other battery, the blue colored which powers the internal RTC of STM32F103 last many years...

@@youtherbiswas8 ok sir, Actually this is just for academic interest? I wanted to make such a clock for practical application, but battery life is too short. Hope you will improve implementation

@@pristineaudio5642 It is a Prototype I made after seeing such worst digital clocks in the market from brands like Ajanta which lose time or gain time & also are not portable as they use a power chord & don't run on batteries..2 days for continuous use with RECHARGEABLE battery, 18650mAh which needs only 30-45 mins to fully recharge...if I increase with one more rechargeable battery, I will get a runtime of 4-5 days continuous use..7-segment displays will consume power...

​@@pristineaudio5642 also NO use of any external RTC chip, internal RTC of STM32F103 is used in this prototype...so biggest advantage is you save money for purchasing External RTC chips like DS1302, DS1307 which adds to your BOM cost...so we solve this problem by using Internal RTC of STM32F103...

FLAC is compressed format, which would need decoding at our ESP-32 Application side, after which we will get the actual PCM samples which can then be written to I2S DMA playback channels to be sent as DOUT to PCM5122 DAC..so YES, it can be played by PCM5122 DAC souncard, but would involve the FLAC decoding logic for FLAC encoded file to be handled in our ESP-32 firmware Application to get PCM samples, 32-bit/16-bit..

Its a Mono Amplifier...check if your stereo is actually dual mono...hehehe...

It should be....as ADPCM samples...16-bit PCM samples recording(audio capture) ...16-bit left plus 16-bit right samples for stereo...Mono recording also is supported....and using ALSA....that would need some extra work in the driver..

No i did not...modify the VS1053 Codec board in anyway....except using the Expansion header pins to interface with the Rpi3...Pins 33 & 34 are already tied...GPIO0 & GPIO1...in the schematic of my VS1053 Codec Module board they are tied via 100k Resistors..forming a Voltage divider between them....please refer my blog page.... blogsmayan.blogspot.com/p/linux-device-driver.html

In my blog page...check the "introduction" section.... I have posted the schematic of my VS1053 Module board...you can see clearly the pins 33 & 34 are tied forming a Voltage Divider circuit between them... blogsmayan.blogspot.com/p/linux-device-driver.html

Please download & check the Linux Driver sources & Device Tree for VS1053 Codec Module board from "Resources Section" of my blog page...

Thanks for the info. I have found a software fix to the lack of tie down on pin 34. I am trying to record with it. I will look at the driver.

I have... check my blog pages..

The WM8960 is a Hardware Audio Codec. It is not responsible for providing streaming protocol support. The RTP Streaming and transcoding will happen at the Master side, the Rpi3 board which is responsible for transcoding it to WAV/MP3 format using FFMPEG or GStreamer and this transcoded stream is just given to the AUDIO SINK SOUND Card or the SLAVE which is your ALSA Driver for WM8960 which handles the playback based on Sampling Rate, no. of channels, the bit clock and bits per sample of L/R(stereo)...

Cannot say that, as TFA9892/TFA9894 AMP is good in some aspects like speaker excursion & protection by means of a dynamic voltage boost control, has a cool flux internal DSP which provides AEC feedback signal, Automatic Gain control, DRC, 10-Band equalizer, Soft MUTE controls to prevent POP noise suppression....so far so good for a Speaker Amplifier...so yes, better than snapdragon 8xx in some aspects...

Hi, Treadstone...the AMP is TFA9892....As per, the Datasheet of TFA9892, "The TFA9892 is a high efficiency class-D audio amplifier with a sophisticated speaker boost and protection algorithm. It can deliver 13.2 W peak output power into an 8 Ohm speaker at a supply voltage of 4.0 V. The internal boost converter raises the supply voltage to 12 V, providing ample headroom for major improvements in sound quality."

For 2nd question, I don't think it can support greater than 32 Ohm....The AMP calibration firmware will fail to program the AMP Chip as per the Load, due to which the Amplifier will not detect any Load speaker driver > 32 Ohm... As, per Datasheet of TFA9892, "The device can be configured to drive either a hands-free speaker (4 Ohm to 8 Ohm) for audio playback, or a receiver speaker (32 Ohm) for handset playback, allowing it to be embedded in platforms that support either or both options. "

For 1st question, 1. Can Tfa98xx run hi-res audio? (24 bit or above)... The answer is YES...I am playing here 16-Bit Stereo Audio, 48 kHz, so 16-Bit Left, 16-Bit Right channel....so 32 bits of Audio sample, as per Datasheet, i can see it supports, till 256-Bit stereo Audio... "number of bit clock pulses (BCK) per sample rate (fs)- 32, 48, 64, 96, 128, 192, 256, 384, 200, 100, 144, 288, 512...." I can see 48-bit Audio sample, 24-bit Stereo is supported....at Playback sampling rates of 16 kHz, 32 kHz, 44.1 kHz and 48 kHz ....as per the TFA9892 Datasheet...please keep in mind the exact TFA98xx...they differ in few aspects...like TFA9894 is different in a few things from TFA9892....etc, etc...

Can u please say Where can I get the datasheet

Good question....I have to add that support in my DRiver...My DRiver right now only supports Audio PLayback...not Audio Capture...my VS1053 Board does have a MIC, which should be very basic on MICP & MICN pins of VS1053 chip..also there is a LineIN 3.5 mm jack for Audio Input/record...In the Datsheet, SCI_MODE Control register, (Add: 0x00) Bit -14 talks about setting MICP/MICN for AUdio record...also bits 12, 14 of SCI_MODE sets the SM_RESET & SM_ADPCM for ADPCM recording mode...check section 9.8.1, ACTIVATING ADPCM Mode...SCI_AICTRL0, for sample rate, SCI_AICTRL1 for Gain, SCI_AICTRL3 for channel setting, Left, right, joint stereo, dual channel....

Also you need to WRITE To WRAM PROGRAM memory, WRAMADDR 0x07..the WriteVS10xxPatch() for setting the DSP Audio Pipeline program & parameters for Audio REcord...check...www.vlsi.fi/en/support/software/vs10xxpatches.html

Also you need to WRITE To WRAM PROGRAM memory, WRAMADDR 0x07..the WriteVS10xxPatch() for setting the DSP Audio Pipeline program & parameters for Audio REcord...check...www.vlsi.fi/en/support/software/vs10xxpatches.html

Good work. I am implemented support for VS 1003 too ..)) But it for user space only..

Yes, I agree Jonathan....the Input Sine Wave Tone is not from a Pure source...the Sine Wave Tone is having other frequencies too...I haven't used any AP, Audio Precision series machine or Sine wave freq. generator source but i downloaded just sine wave tone .wav files from some Internet source which are not quite preferred way for Audio testing...just to have a feel of how the major freq. would look on the spectrum and was satisfied with it, as i knew the Input source was not a pure source but having other frequencies too...I knew the Input was not perfect, hence there would be other frequencies too that would be showing up, but was happy to see the band around the lower end have more depth than the higher when I gave a Low freq Sine wave input, similarly the Mid Band had more depth when gave a Mid Freq Sine wave and so with High band when a High freq Sine wave was input...that was conclusive that the Spectrum was correct...as per the Imperfections in the Input, which was again not 100% pure...

Hi Jonathan.....Yes, You are right, Jonathan....Highs are on the LEFT of the Display, around the 10kHz band, the MID Section of the Display is for the Mids, 630Hz, 1kHz , 3.3kHz till 6.3kHz and the Lows are at the RIGHT of the Display, starting from 50Hz, 100Hz, 150Hz, 250Hz till 630Hz.....Here's my Sine wave Tone Test Video, Jonathan which clearly shows the distribution...I have used Sine Wave tone frequencies of 50Hz, 100Hz, 250Hz, 440Hz, 630Hz, 1kHz, 2Khz, 6.3khz and 10kHz in this demonstration.....Here's the Link... ruclips.net/video/O0S8s92vqiI/видео.html