Audio Core Changelog¶

v0.7.0 - 2022-11-30¶

Release notes:

The fallback module has been optimized to work better with the Wireless Core’s clear channel assessment (CCA) feature.
The CDC module module/audio_cdc_module.c/h has been removed and converted to a regular processing stage processing/sac_cdc.c/h.
All endpoints, modules and processing stages prefix has been changed from audio_ to sac_.
Statistics functions have been moved from sac_api.c/h to sac_stats.c/h.
The Audio Core API reference section in the documentation has been enhanced.
The Audio Core Fallback section in the documentation has been enhanced.
A CDC processing stage using clock steering as compensation mechanism procesing/sac_cdc_clock_steering.c/h is now available.

Added

Fallback module has the new get_cca_metrics public function.
New cca_max_try_count member in sac_fallback_module_cfg_t.
New sac_cca_metrics_t type in sac_api.h.
New cca_metrics member in sac_fallback_module_t.
SAC_LINK_MARGIN_SIZE_BYTES constant in sac_api.h.
cdc_inflated_packets_count and cdc_deflated_packets_count counters in sac_statistics_t.
New API call sac_pipeline_get_cdc_inflated_packets_count().
New API call sac_pipeline_get_cdc_deflated_packets_count().
New sac_stats.c/h files.
New API call sac_pipeline_update_stats().
New API call sac_pipeline_format_stats().
New consumer_queue_peak_buffer_load, consumer_link_margin_min_peak, consumer_cca_fail_count_peak statistics and getters.

Changed

fallback_set_rx_link_margin parameter is now a uint8_t``instead of ``int16_t.
fallback_update_stats now has the cca_fail_count parameter.
In sac_link_margin_metrics_t, link_margin_accumulator is now a uint32_t instead of a int32_t and link_margin_acc_avg is now a uint8_t instead of a int16_t.
In sac_fallback_module_t, fallback_threshold, fallback_threshold_default and fallback_threshold_hysteresis are now a uint8_t instead of a uint16_t. Also, rx_link_margin_data_rx is now a uint8_t instead of a int16_t.
Replace AUDIO_ prefix to SAC_ for sac_bit_depth_t and sac_word_size_t members.
Rename audio_endpoint_cfg to sac_endpoint_cfg.
Rename sac_mixer_module_init to sac_mixer_init.
Rename sac_fallback_module_init to sac_fallback_init.
Rename fallback_* functions to sac_fallback_*.

Removed

Remove compression_enabled field from sac_compression_instance_t as well as SAC_COMPRESSION_ENABLE and SAC_COMPRESSION_DISABLE from sac_compression_cmd_t. Such control should be implemented by the user in the processing stage gate() function.
Remove packing_enabled field from sac_packing_instance_t as well as SAC_PACKING_ENABLE and SAC_PACKING_DISABLE from sac_packing_cmd_t. Such control should be implemented by the user in the processing stage gate() function.

Fixed

Added

The Audio Core now supports 20, 24 and 32-bit audio samples.
New processing stage that allows packing of 20 or 24-bit audio samples from 32-bit words.
New sinus generator endpoint that can produce pre-recorded sin waves at a frequency of 1 kHz, 2 kHz and 3 kHz for testing purposes.
New audio mixing module used for mixing up to 3 audio streams into a single one.
audio_protocol_cfg_t now has a sac_mixer_option_t in order to select if the pipeline is producing an audio stream to be mixed or if the pipeline is doing the mixing.
New API call sac_mixer_module_init() used to initialize the audio mixer module (if used).
New API call sac_endpoint_link() used to make 2 endpoints share their audio queue (for use with the audio mixing feature).
New API call sac_pipeline_add_extra_producer() used to add more than one audio producer in a pipeline (for use with the audio mixing feature).
New SAC_ERR_NULL_PTR error that happens when a pointer is NULL while it should have been initialized.
New SAC_ERR_MIXER_INIT_FAILURE error that happens when the mixer is not able to initialize properly due to an erroneous configuration.
New audio fallback module.
New API call sac_fallback_module_init() used to initialize the audio fallback module (if used).
New producer_packets_corrupted_count stat.
New CRC4 field in audio header used to validate the content of the audio header.
New gate function pointer field in the Audio Processing Interface (sac_processing_interface_t).

Changed

Prefix of the SPARK Audio Core (SAC) API changed from audio_ to sac_.
audio_protocol.c/h has been renamed to audio_cdc_module.c/h and moved to the module folder.
audio_protocol_cfg_t has been renamed to sac_pipeline_cfg_t.
Audio core now supports payload size up to 65535 bytes in its pipelines.
Audio payload is now aligned with 32-bit words.
Audio volume processing stage now has the prefix audio_ on all public functions.
The audio processing stage process function now returns a uint16_t payload size.
The bit depth field of the endpoint config structure now has a typedef enum to list all supported bit depths.
The payload size field of the endpoint config structure now has a uint16_t value.
sac_pipeline_process() now mixes audio packets when the audio mixing feature is enabled.

Added

Audio buffer load check inside audio_pipeline_consume().
A new field in audio_protocol_cfg_t has been added to turn on or off the user data feature. The audio packet trailing byte will not be appended if it is turned off.

Fixed

Added

New API call audio_processing_stage_init() that initializes a processing stage and returns a reference to it.
New API call audio_pipeline_setup() used to do the final setup of the pipeline before it is operational.
New API call audio_core_get_allocated_bytes() used to know the number of bytes used in the memory pool.
New AUDIO_ERR_PIPELINE_CFG_INVALID error that happens when incompatible pipeline configurations are set.

Changed

audio_pipeline_init() now dynamically allocated the pipeline and returns a reference to it.
audio_endpoint_init() now dynamically allocated the endpoint and returns a reference to it.
Producer endpoint audio_payload_size field needs not to account for audio packet size inflation during subsequent audio processing like in the case of upsampling or decompression. The value must always be set to the exact amount of bytes the produced audio packet is.
The block_size field of the src_cmsis_cfg_t type of the Sampling Rate Converter processing stage has been replace by a payload_size field and a bit_depth field.
A const qualifier has been added to the name field of the audio_processing_t type.
The max_process_count field of the audio_protocol_cfg_t has been removed. Processing stages are allocated dynamically using the chunk of RAM allocated by the application.
Error pointer parameter has been added to audio_endpoint_init() and removed from audio_pipeline_add_processing().
audio_pipeline_stop() no longer de-initializes processing stages.

Removed

WPS audio endpoint. Audio applications must supply their own. A working implementation is provided with the SDK.

Added

Changed

Removed

Generic I2S audio endpoint. Audio applications can create their own specific audio endpoints if needed instead.

Fixed

Changed

Sampling rate converter processing stage filter coefficients when interpolating.

Initial release.