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Notice: Pre-Release Documentation
This document is part of a prerelease and is currently a work in progress. Some content may be incomplete, subject to change, or marked as TBD. We are actively updating this documentation and will continue to provide the most accurate and up-to-date information as development progresses. Thank you for your understanding!

This section contains some performance benchmarks for this release. During the measurements, processor clock and DDR speeds were set to maximum.

Interprocessor Communication

This is the measured MHz to transfer 16 channels of 32-bit data at 48 kHz between cores. Transfers were done with 1 msec blocks. This test uses the ChangeThread module and part of the reported MHz is on the sending core and part is on the receiving core.

...

When sending data between different cores, it is going through the carveout shared memory. When sending audio between the same core, this means that data is going to another hardware thread on the same core and non-shared is used.

TDM I/O

The limiting factor for serial port I/O is the speed of the LP DMA memory. This is 85 MB/second for reading and 170 MB/second for writing. These tests were with 48 sample block sizes.

...

Block Size (samples)

Total

Latency

(msec)

Analog

Latency

(msec)

Digital

Latency

(msec)

12

1.6

1.1

0.5

24

2.1

1.1

1.0

48

3.1

1.1

2.0

...

ALSA I/O

This test measures the overhead of streaming audio data between the HLOS and Audio Weaver. 16 channels of data streamed at 48 kHz. 1 msec block size in Audio Weaver and a 10 msec block size at the HLOS.

...

An application was running on the HLOS which would read the ALSA Source and send it to the ALSA Sink. A 48 kHz sample rate was used throughout and Audio Weaver was processing at a 48 sample block size. The HLOS application was using a 240 sample block size. The ALSA settings used in the test were:

ALSA Sink

  • Buffer size: 960 samples

  • Block size: 240 samples

  • startThreshold: 0 samples

  • stopThreshold: 0 samples

ALSA Source

  • Buffer size: 960 samples

  • Block size: 240 samples

  • startThreshold: 480 samples (prefill to avoid underruns)

  • stopThreshold: 0 samples

...

ALSA Settings

Measured

Sample Rate

Block Size

Buffer Size

Latency (msec)

8000

80

160

18.2

11025

120

240

21.6

12000

120

240

21.8

16000

160

320

15.6

22050

220

440

20.0

24000

240

480

15.4

32000

320

640

17.8

44100

480

960

19.8

48000

480

960

16.2

Maximum CPU Loading

In this test, we used the BiquadLoading module to load each thread in the system. We measured how many Biquad stages we could run before we started having CPU overruns. We used a 1 msec block size on the Hexagon DSPs and a 10 msec block size on the Arm. This test measures code and framework efficiency.

ADSP

ThreadBiquadStages

Biquad Stages

% Loading

1A

1600

95%

1B

1720

95%

1C

1720

95%

1D

1720

95%

GPDSP0

ThreadBiquadStages

Biquad Stages

% Loading

1A

2000

90%

1B

2000

90%

1C

2000

90%

1D

2000

90%

1E

2000

90%

1F

2000

90%

GPDSP1

ThreadBiquadStages

Biquad Stages

% Loading

1A

2000

90%

1B

2000

90%

1C

2000

90%

1D

2000

90%

1E

2000

90%

1F

2000

90%

Arm (only loaded a single thread)

ThreadBiquadStages

Biquad Stages

% Loading

10A

5000

50%

Early Audio KPIs

In release R4.0, we are not able to fully measure this KPI. However, we were able to measure the Audio Weaver contribution to the boot time. We instrumented the code and measured the time that the main() function was reached on the ADSP until the time that real-time audio interrupts started.

...

The time from the ADSP booting until GPDSP0 and GPDSP1 are ready to generate audio is 27.5 msec.

...

A2B Board Setup

The board shown below is the “Rev D” version. There are 4 smaller jumper switches and they need to be set as shown:

...

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1 Unless, of course, you want to write your own custom modules that execute on the Hexagon.

5 This is how varying sample rates are handled in Audio Weaver. The buffers are oversized for the worst case transfer size and side information - in the timing information pins - is used to regulate the flow.

6 The discussion here is based on block times which are easier to follow. The actual configuration is based on block sizes.

4 This is a new feature which was recently implemented in Audio Weaver. If your design has custom modules, then you may need to add a custom resetState function.

2 The objectID of this SourceInt module is specified in the system configuration file.

3 You will also need a Matlab license. We recommend Matlab 2022b.