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[mtl-007] AEC fixes from main branch submission #8603

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[mtl-007] AEC fixes from main branch submission #8603

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43ff9f7
Revert "[DO NOT UPSTREAM] re-introduce accidently removed bu size code"
andyross Jan 3, 2024
d206a13
Revert "module: Add support for saving and retrieving the base config…
andyross Jan 3, 2024
e639237
zephyr/alloc: Add newlib-style allocator stub for C++ builds
andyross Mar 2, 2023
0285502
zephyr: Update GOOGLE_RTC_AUDIO_PROCESSING
andyross Mar 2, 2023
fb33be3
audio_stream: Fix uninitialized data in alignment_constants
andyross Dec 23, 2023
928156a
various components: Remove default/stub init_alignment_constants usage
andyross Dec 23, 2023
6fd333e
module_adapter_ipc4: Save and pre-parse base_cfg_ext data
andyross Jan 2, 2024
af520b4
ipc4: Use saved base_cfg_ext data in ipc_comp_connect()
andyross Jan 3, 2024
3014415
google_rtc_audio_processing: Fixes for MTL branch
andyross Dec 22, 2023
511dbaf
[DNM] Google AEC: minimize workaround
andyross Jan 3, 2024
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4 changes: 1 addition & 3 deletions src/audio/aria/aria.c
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Original file line number Diff line number Diff line change
Expand Up @@ -162,9 +162,7 @@ static void aria_set_stream_params(struct comp_buffer *buffer,
const struct ipc4_audio_format *audio_fmt = &mod->priv.cfg.base_cfg.audio_fmt;

ipc4_update_buffer_format(buffer, audio_fmt);
#ifdef ARIA_GENERIC
audio_stream_init_alignment_constants(1, 1, &buffer->stream);
#else
#ifndef ARIA_GENERIC
audio_stream_init_alignment_constants(8, 1, &buffer->stream);
#endif
}
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3 changes: 0 additions & 3 deletions src/audio/asrc/asrc.c
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Expand Up @@ -755,9 +755,6 @@ static int asrc_prepare(struct processing_module *mod,
sinkb = list_first_item(&dev->bsink_list,
struct comp_buffer, source_list);

audio_stream_init_alignment_constants(1, 1, &sourceb->stream);
audio_stream_init_alignment_constants(1, 1, &sinkb->stream);

/* get source data format and period bytes */
cd->source_format = audio_stream_get_frm_fmt(&sourceb->stream);
source_period_bytes = audio_stream_period_bytes(&sourceb->stream,
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1 change: 1 addition & 0 deletions src/audio/audio_stream.c
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Expand Up @@ -141,6 +141,7 @@ void audio_stream_init(struct audio_stream *audio_stream, void *buff_addr, uint3
audio_stream->addr = buff_addr;
audio_stream->end_addr = (char *)audio_stream->addr + size;

audio_stream_init_alignment_constants(1, 1, audio_stream);
source_init(audio_stream_get_source(audio_stream), &audio_stream_source_ops,
&audio_stream->runtime_stream_params);
sink_init(audio_stream_get_sink(audio_stream), &audio_stream_sink_ops,
Expand Down
5 changes: 1 addition & 4 deletions src/audio/crossover/crossover.c
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Expand Up @@ -733,14 +733,11 @@ static int crossover_prepare(struct processing_module *mod,
/* Get source data format */
cd->source_format = audio_stream_get_frm_fmt(&source->stream);
channels = audio_stream_get_channels(&source->stream);
audio_stream_init_alignment_constants(1, 1, &source->stream);

/* Validate frame format and buffer size of sinks */
list_for_item(sink_list, &dev->bsink_list) {
sink = container_of(sink_list, struct comp_buffer, source_list);
if (cd->source_format == audio_stream_get_frm_fmt(&sink->stream)) {
audio_stream_init_alignment_constants(1, 1, &sink->stream);
} else {
if (cd->source_format != audio_stream_get_frm_fmt(&sink->stream)) {
comp_err(dev, "crossover_prepare(): Source fmt %d and sink fmt %d are different.",
cd->source_format, audio_stream_get_frm_fmt(&sink->stream));
ret = -EINVAL;
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13 changes: 0 additions & 13 deletions src/audio/dcblock/dcblock.c
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Expand Up @@ -206,17 +206,6 @@ static int dcblock_process(struct processing_module *mod,
return 0;
}

/* init and calculate the aligned setting for available frames and free frames retrieve*/
static inline void dcblock_set_frame_alignment(struct audio_stream *source,
struct audio_stream *sink)
{
const uint32_t byte_align = 1;
const uint32_t frame_align_req = 1;

audio_stream_init_alignment_constants(byte_align, frame_align_req, source);
audio_stream_init_alignment_constants(byte_align, frame_align_req, sink);
}

#if CONFIG_IPC_MAJOR_4
static void dcblock_params(struct processing_module *mod)
{
Expand Down Expand Up @@ -266,8 +255,6 @@ static int dcblock_prepare(struct processing_module *mod,
/* get sink data format and period bytes */
cd->sink_format = audio_stream_get_frm_fmt(&sinkb->stream);

dcblock_set_frame_alignment(&sourceb->stream, &sinkb->stream);

dcblock_init_state(cd);
cd->dcblock_func = dcblock_find_func(cd->source_format);
if (!cd->dcblock_func) {
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9 changes: 0 additions & 9 deletions src/audio/drc/drc.c
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Expand Up @@ -227,14 +227,6 @@ static int drc_get_config(struct processing_module *mod,
return comp_data_blob_get_cmd(cd->model_handler, cdata, fragment_size);
}

static void drc_set_alignment(struct audio_stream *source,
struct audio_stream *sink)
{
/* Currently no optimizations those would use wider loads and stores */
audio_stream_init_alignment_constants(1, 1, source);
audio_stream_init_alignment_constants(1, 1, sink);
}

static int drc_process(struct processing_module *mod,
struct input_stream_buffer *input_buffers,
int num_input_buffers,
Expand Down Expand Up @@ -308,7 +300,6 @@ static int drc_prepare(struct processing_module *mod,
/* DRC component will only ever have 1 source and 1 sink buffer */
sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, sink_list);
sinkb = list_first_item(&dev->bsink_list, struct comp_buffer, source_list);
drc_set_alignment(&sourceb->stream, &sinkb->stream);

/* get source data format */
cd->source_format = audio_stream_get_frm_fmt(&sourceb->stream);
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24 changes: 14 additions & 10 deletions src/audio/google/Kconfig
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Expand Up @@ -15,8 +15,6 @@ config COMP_GOOGLE_HOTWORD_DETECT
config COMP_GOOGLE_RTC_AUDIO_PROCESSING
bool "Google Real Time Communication Audio processing"
select COMP_BLOB
select GOOGLE_RTC_AUDIO_PROCESSING_MOCK if COMP_STUBS
depends on IPC_MAJOR_4
default n
help
Select for Google real-time communication audio processing. It
Expand All @@ -25,13 +23,7 @@ config COMP_GOOGLE_RTC_AUDIO_PROCESSING
This component takes raw microphones input and playback reference
and outputs an echo-free microphone signal.

config COMP_GOOGLE_RTC_USE_32_BIT_FLOAT_API
depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
bool "Use 32bit API in Google Audio processing"
default n
help
Selects an API to be used in communication with the Google real-time
communication audio processing: 32bit float or 16bit integer
if COMP_GOOGLE_RTC_AUDIO_PROCESSING

config COMP_GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ
depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
Expand All @@ -41,6 +33,16 @@ config COMP_GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ
Sets the sample rate for the memory buffer for the Google real-time
communication audio processing.

config COMP_GOOGLE_RTC_AUDIO_PROCESSING_CHANNEL_MAX
int "Max number of AEC channels"
default 2
help
Sets the maximum number source/sink channels Google Real
Time Communication Audio Processing will use for. This is a
computation and memory budget tunable. Channel counts are
retrieved at runtime, but channels higher than this number
are ignored (on input) or cleared (output).

config COMP_GOOGLE_RTC_AUDIO_PROCESSING_MEMORY_BUFFER_SIZE_BYTES
depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
int "Memory buffer size for Google Real Time Communication Audio processing"
Expand All @@ -67,10 +69,12 @@ config COMP_GOOGLE_RTC_AUDIO_PROCESSING_MIC_HEADROOM_LINEAR

config GOOGLE_RTC_AUDIO_PROCESSING_MOCK
bool "Google Real Time Communication Audio processing mock"
default n
default y if COMP_STUBS
depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
help
Mock Google real-time communication audio processing.
It allows for compilation check and basic audio flow checking.

endif # COMP_GOOGLE_RTC_AUDIO_PROCESSING

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