Audio Bidirectional

Description

The Audio Bidirectional application uses the SPARK Wireless Core to transmit a stereo audio stream with 48kHz/24-bit depth from the Coordinator to the Node (the Main Channel), and conversely, a mono audio stream with 32kHz/16-bit depth from the Node back to the Coordinator (the Back Channel). When the link quality degrades, it will automatically switch to fallback mode which involves a compressed audio stream on both sides. The audio stream will revert to uncompressed once the link quality returns to normal.

The application can either use the analog audio codec or the USB audio interface to receive and transmit audio data. Different compilation flags are required to select the desired audio interface. A USB audio Coordinator can communicate with an analog audio Node and vice versa.

To compile the application for USB audio, set the “USB_AUDIO_ENABLED”: “ON” flag in the application’s CMake preset cacheVariables section.

     {
         "name": "audio-bidirectional-sr11xx-quasar-usb_audio",
         "inherits": ["base","quasar"],
         "description": "Audio bidirectional example for SR11xx with Quasar USB AUDIO",
         "cacheVariables": {
             "APP": "Audio-Bidirectional",
             "TRANSCEIVER": "SR1100",
             "USB_AUDIO_ENABLED": "ON"
         }
     },

Behavior

Audio Configuration

The application can be configured using the sac_cfg.h file located in the config folder.

/* **** Main Channel Settings. **** */

#define MAIN_CHANNEL_SAMPLE_RATE_HZ 48000
#define MAIN_CHANNEL_SAMPLE_COUNT   20
#define MAIN_CHANNEL_CHANNEL_COUNT  2
#define MAIN_CHANNEL_BIT_DEPTH      24
#define MAIN_CHANNEL_LATENCY_MS     5

/* Calculated values. */
#define MAIN_CHANNEL_SWC_PAYLOAD_SIZE \
    SAC_CALCULATE_PAYLOAD_SIZE(MAIN_CHANNEL_SAMPLE_COUNT, MAIN_CHANNEL_CHANNEL_COUNT, MAIN_CHANNEL_BIT_DEPTH)
#define MAIN_CHANNEL_I2S_PAYLOAD_SIZE \
    SAC_CALCULATE_PAYLOAD_SIZE(MAIN_CHANNEL_SAMPLE_COUNT, MAIN_CHANNEL_CHANNEL_COUNT, I2S_DMA_BIT_DEPTH)
/* Size of the latency queue used by the Audio Core for the main channel. */
#define MAIN_CHANNEL_LATENCY_QUEUE_SIZE                                                                    \
    SAC_CALCULATE_LATENCY_QUEUE_SIZE(MAIN_CHANNEL_LATENCY_MS, CODEC_LATENCY_MS, MAIN_CHANNEL_SAMPLE_COUNT, \
                                     MAIN_CHANNEL_SAMPLE_RATE_HZ)

/* **** Back Channel Settings. **** */

#define BACK_CHANNEL_SAMPLE_RATE_HZ 32000
#define BACK_CHANNEL_SAMPLE_COUNT   52
#define BACK_CHANNEL_CHANNEL_COUNT  1
#define BACK_CHANNEL_BIT_DEPTH      16
#define BACK_CHANNEL_LATENCY_MS     15

/* Calculated values. */
#define BACK_CHANNEL_SWC_PAYLOAD_SIZE \
    SAC_CALCULATE_PAYLOAD_SIZE(BACK_CHANNEL_SAMPLE_COUNT, BACK_CHANNEL_CHANNEL_COUNT, BACK_CHANNEL_BIT_DEPTH)
#define BACK_CHANNEL_I2S_SAMPLE_COUNT ((BACK_CHANNEL_SAMPLE_COUNT * I2S_SAMPLE_RATE_HZ) / BACK_CHANNEL_SAMPLE_RATE_HZ)
#define BACK_CHANNEL_I2S_PAYLOAD_SIZE \
    SAC_CALCULATE_PAYLOAD_SIZE(BACK_CHANNEL_I2S_SAMPLE_COUNT, BACK_CHANNEL_CHANNEL_COUNT, I2S_DMA_BIT_DEPTH)
/* Size of the latency queue used by the Audio Core for the back channel. */
#define BACK_CHANNEL_LATENCY_QUEUE_SIZE                                                                    \
    SAC_CALCULATE_LATENCY_QUEUE_SIZE(BACK_CHANNEL_LATENCY_MS, CODEC_LATENCY_MS, BACK_CHANNEL_SAMPLE_COUNT, \
                                     BACK_CHANNEL_SAMPLE_RATE_HZ)

Audio Interface

I2S Audio Interface

The application receives audio from the line in of either the Coordinator or the Node, utilizes the audio codec to transmit it over the I2S line using DMA transfers, and outputs the audio to the headphone out of the opposite device.

The audio core pipeline processes audio samples coming from the I2S to format them to be sent over the air using the SWC. The I2S end of transfer callback and the SWC transmission success callback will trigger the audio processing to synchronize the I2S production rate and the SWC consumption rate.

Figure 5: Audio application timings in normal operation.

The Audio Task row shows when swc_connection_send() actually enqueues the packet. The event that decides which slot the packet catches is the connection’s frame preparation (prepare_frame), which selects and latches the payload for the next assigned slot. Frame preparation runs in the SPI DMA ISR context (see the prepare_frame step in the Wireless Core timing diagram).

Each transmit slot first dequeues the previously acknowledged packet, then pulls the next one from the SWC queue to prepare it, so the queue must be able to hold at least two entries to track both the in-flight and the next packet.

If the Audio Task enqueues after that preparation point, the packet misses the slot, which ships empty, and it must wait for the following preparation, adding one inter-opportunity gap to its wireless latency; see Wireless Latency for the corresponding formulas.

When a packet transmission fails, the audio packets will accumulate in the audio consumer queue until the wireless core can transmit its payloads and free a space in its queue.

Figure 6: Audio application timings when a packet transmission fails.

USB Audio Interface

The application receives audio from the USB audio interface at either 1 kHz (USB Full-Speed) or 8 kHz (USB High-Speed) depending on the capabilities of the device.

Note

When using USB audio, the host computer controls the playback. If audio playback stops on the host computer, no audio samples are receives on the device. This means that the audio connection will not send any packets.

These frames are then accumulated and splitted into audio packets to be sent to the audio pipeline for processing before being sent OTA using the SWC. When using The USB interface, the audio samples are already formatted to packed 24-bit.

On the receiving side, the audio packets received OTA are processed and sent to the USB audio interface where a FIFO will accumulate the samples.

Note

When using USB audio, the host computer controls the recording. If the audio recording stops on the host computer, no audio samples are read from the device. This means that the USB audio transmission FIFO will not be depleted and will eventually overflow. Once the recording resumes, the FIFO will be consumed again and will stabilize to the correct level to match the expected latency.

The bidirectional link also allows for the transmission of various data such as the status of a button and the link margin. The link margin makes it possible for the Coordinator to track the quality of the connection, thereby switching between fallback and normal modes as needed. This data is transmitted when the wireless is free. The transmission and reception of data are assigned the lowest priority, prioritizing audio transfers while still being capable of transferring this data.

Note

The Coordinator’s data connection is part of the synchronization design of this application. With USB audio, the audio connection stops transmitting entirely while the host is not streaming, and the Coordinator’s periodic data transmission becomes the Node’s only synchronization source, acting as the network beacon. Its period must not exceed 10 ms so that two data frames fit within the Node’s synchronization-loss timeout, tolerating one lost frame (see Sync Timeslot (Beacon)). Removing this data connection or increasing its period would cause the Node to repeatedly lose synchronization while the audio stream is idle.

The wireless schedule allocates 7 timeslots for the transmissions from the Coordinator and 2 timeslots for its reception. The schedule is designed to reserve more bandwidth for transmitting a higher-quality audio stream from the Coordinator to the Node.

In the UWB frequency range, the wireless core alternates between 4 different frequency bands at each transmission.

Application Schedule

The following diagram illustrates the general structure of the schedule used in this application.

Audio Bidirectionnal schedule — Figure 7: Audio Bidirectional Application’s Schedule.

Pairing

On boot, devices attempt to pair automatically. If unsuccessful, manual pairing can be performed afterward to use the application.

Auto Pairing:

On boot, devices automatically start the pairing process. RGB blinks blue twice to indicate that pairing is in progress.
If the pairing procedure is unsuccessful, RGB blinks red twice after 10 seconds.
Once successfully paired, RGB turns magenta, and the devices begin exchanging application data.

Manual Pairing:

Press the SW1 button on both devices to start the pairing procedure; RGB blinks blue twice when doing so.
If the pairing procedure is unsuccessful, RGB blinks red twice after 10 seconds.
Once successfully paired, RGB turns magenta, and the devices begin exchanging application data.

Unpairing:

When paired, pressing the SW1 button again unpairs the device; RGB blinks red twice to indicate that unpairing has been completed.
Unpairing causes the devices to forget their learned addresses. The devices are no longer able to exchange application data.
To start a new pairing procedure, both devices must be unpaired first.

Fallback

The transmitter continuously monitors the link quality. If the link quality degrades, the system will enter fallback mode to compress audio packets on both sides. It employs the ADPCM algorithm for audio data compression. This compression algorithm reduces audio samples by a factor of 4. This stage helps lower the application’s throughput to increase range while still ensuring relatively good audio quality. While in fallback mode, if the link quality remains strong for 5 seconds and the CCA try count stays below a defined threshold, the system will revert to normal mode. The thresholds for entering and exiting fallback mode differ, creating hysteresis to prevent the system from toggling frequently in conditions of marginal link quality.

Clock Drift Compensation (CDC)

Clock Drift Compensation (CDC) is used to adjust the timing difference between the two devices. The Coordinator is the reference clock. Therefore, for the stereo audio Main Channel stream, the Node can use its internal PLL (when hardware allows) to adjust its clock frequency to the Coordinator’s. For the mono audio Back Channel stream, the Coordinator cannot modify its clock since it is the reference. CDC is therefore performed by using software resampling. For more information regarding CDC, refer to Clock Drift Compensation.

Board User Interface

The board user interfaces are identical for the Node and the Coordinator and includes LEDs, RGB and button switches.

Table 1: Quasar Board User Interface Description
Interface	Name	Description
SW1	Pairing process	Pair, unpair, or abort the pairing process depending on the current state.
SW2	Button activity	Turn on the LED state of the remote device.
SW3	Volume up	Raises the received audio volume (default at 100%).
SW4	Volume down	Lowers the received audio volume.
LED1	Transmission success	Toggles every time an audio packet is successfully transmitted.
LED2	Reception success	Toggles every time an audio packet is successfully received.
LED3	Fallback status	LED is OFF in Normal mode and ON in Fallback mode.
LED4	Button state	Turn on upon receiving button state.
RGB	Pairing Event	Indicate pairing event as described in the pairing section.

Note

LEDs assigned to transmission or reception status are almost always on because of the high packet rate.
Audio volume can be adjusted by the receiving device, meaning both the Coordinator and Node control the output gain of their respective streams. This control is managed through the SW3 and SW4 buttons.

Certification

This application has the ability to run in Certification Mode.

When certification mode is active, a dedicated LED is activated to indicate that the mode is enabled. All activity status LEDs (transmission, reception, fallback, and payload) are suppressed so that only the certification indicator is visible.

Table 2: Certification Mode Indicator
Board	Indicator
Quasar	RGB LED steady yellow.
EVK1.4	Steady LED2.

Different levels of certification are required to test the fallback modes and data connection.

Depending of which mode is requested, the max_payload_size of the audio connection will be changed.
The order in which connection are initialized and their priority will be swapped if the data connection needs to be tested.

Activating the Certification Mode

The Coordinator and Node have distinct certification modes because they transmit different audio streams: the Coordinator sends the 48 kHz/24-bit main channel and the Node sends the 32 kHz/16-bit back channel. Each certification mode must be activated independently on its own board.

Coordinator

Table 3: Certification Mode Activation for the Coordinator
Certification Mode	Activation Process
48 kHz/24-bit Audio	Hold SW2 at startup until the LED 1 blinks once, then release.
48 kHz ADPCM Audio	Hold SW2 at startup until the LED 1 blinks twice, then release.
Control Data	Hold SW2 at startup until the LED 1 blinks three times, then release.

Node

Table 4: Certification Mode Activation for the Node
Certification Mode	Activation Process
32 kHz/16-bit Audio	Hold SW2 at startup until the LED 1 blinks once, then release.
32 kHz ADPCM Audio	Hold SW2 at startup until the LED 1 blinks twice, then release.
Control Data	Hold SW2 at startup until the LED 1 blinks three times, then release.

Terminal

The following figure shows the terminal output of this application.

See Viewing the Terminal for details on how to view the terminal output.

Audio Bidirectional Terminal Output Audio Core — Figure 8: Audio Bidirectional Terminal Output - Audio Core (from left: Coordinator & Node).

Audio Bidirectional Terminal Output Wireless Core — Figure 9: Audio Bidirectional Terminal Output - Wireless Core (from left: Coordinator & Node).

See Audio Core API - Statistics and Wireless Core API - Statistics for more details.

Latency Breakdown

The following diagrams show the typical end-to-end audio latency for each direction (I2S interface, normal mode). Block widths are proportional to their contribution.

Main Channel (Coordinator to Node) — 48 kHz stereo 24-bit:

Audio Bidirectional Main Channel Latency Breakdown — Figure 10: Main Channel Latency Breakdown.

Table 5: Main Channel Latency
Component	Duration	Notes
TX Codec (ADC)	0.8 ms	MAX98091 ADC path delay at 48 kHz
Capture (DMA)	0.42 ms	1 packet = 20 samples at 48 kHz
Wireless transit (typical)	0.33 ms	E[wait]=181 us, CCA=3 us, airtime=75.5 us, processing=67 us
RX Queue	2.5 ms	6 packets buffered for jitter absorption
RX Codec (DAC)	0.76 ms	MAX98091 DAC path delay at 48 kHz
Total (typical)	~4.8 ms	Target: 5 ms

Back Channel (Node to Coordinator) — 32 kHz mono 16-bit:

Audio Bidirectional Back Channel Latency Breakdown — Figure 11: Back Channel Latency Breakdown.

Table 6: Back Channel Latency
Component	Duration	Notes
TX Codec (ADC)	0.8 ms	MAX98091 ADC path delay at 48 kHz
Capture (DMA)	1.63 ms	1 packet = 78 samples at 48 kHz (I2S rate)
SRC TX (downsample)	0.5 ms	FIR 24-tap filter at 48 kHz input (48 kHz to 32 kHz)
Wireless transit (typical)	0.71 ms	E[wait]=569 us, 2 TX slots per 2.25 ms frame, airtime=67.7 us
RX Queue	8.13 ms	5 packets buffered for jitter absorption
SRC RX (upsample)	0.75 ms	FIR 24-tap filter at 32 kHz input (32 kHz to 48 kHz)
RX Codec (DAC)	0.76 ms	MAX98091 DAC path delay at 48 kHz
Total (typical)	~13.3 ms	Target: 15 ms

Schedule: 9 timeslots of 250 us (2.25 ms frame). Main: 7 TX slots (shared audio + data via priority), gap pattern {250, 250, 500, 250, 250, 250, 500} us. Back: 2 TX slots, gap pattern {1250, 1000} us.

See Wireless Latency and Audio Latency for the formulas used in this breakdown.