Audio: Level multiplier: Add new component

scripts/sof-rebuild-processing-comp-blobs.sh

@@ -41,3 +41,4 @@ cd "$SOF_WORKSPACE"/sof/src/audio/multiband_drc/tune; $OCTAVE sof_example_multib
 cd "$SOF_WORKSPACE"/sof/src/audio/tdfb/tune; ./sof_example_all.sh
 cd "$SOF_WORKSPACE"/sof/src/audio/selector/tune; $OCTAVE ./sof_selector_blobs.m
 cd "$SOF_WORKSPACE"/sof/tools/tune/mfcc; $OCTAVE setup_mfcc.m
+cd "$SOF_WORKSPACE"/sof/src/audio/level_multiplier/tune; $OCTAVE sof_level_multiplier_blobs.m

src/arch/host/configs/library_defconfig

@@ -8,6 +8,7 @@ CONFIG_COMP_GOOGLE_RTC_AUDIO_PROCESSING=y
 CONFIG_COMP_GOOGLE_CTC_AUDIO_PROCESSING=y
 CONFIG_COMP_IIR=y
 CONFIG_COMP_IGO_NR=y
+CONFIG_COMP_LEVEL_MULTIPLIER=y
 CONFIG_COMP_MFCC=y
 CONFIG_COMP_MODULE_ADAPTER=y
 CONFIG_COMP_MULTIBAND_DRC=y

src/audio/CMakeLists.txt

@@ -123,6 +123,9 @@ if(NOT CONFIG_COMP_MODULE_SHARED_LIBRARY_BUILD)
 	if(CONFIG_COMP_TEMPLATE_COMP)
 		add_subdirectory(template_comp)
 	endif()
+	if(CONFIG_COMP_LEVEL_MULTIPLIER)
+		add_subdirectory(level_multiplier)
+	endif()
 endif()
 
 ### Common files (also used in shared library build)

src/audio/Kconfig

@@ -180,6 +180,8 @@ rsource "codec/Kconfig"
 
 rsource "template_comp/Kconfig"
 
+rsource "level_multiplier/Kconfig"
+
 endmenu # "Audio components"
 
 menu "Data formats"

src/audio/level_multiplier/CMakeLists.txt

@@ -0,0 +1,15 @@
+# SPDX-License-Identifier: BSD-3-Clause
+
+if(CONFIG_COMP_LEVEL_MULTIPLIER STREQUAL "m")
+  add_subdirectory(llext ${PROJECT_BINARY_DIR}/level_multiplier_llext)
+  add_dependencies(app level_multiplier)
+else()
+  add_local_sources(sof level_multiplier.c)
+  add_local_sources(sof level_multiplier-generic.c)
+  add_local_sources(sof level_multiplier-hifi3.c)
+  add_local_sources(sof level_multiplier-hifi5.c)
+
+  if(CONFIG_IPC_MAJOR_4)
+    add_local_sources(sof level_multiplier-ipc4.c)
+  endif()
+endif()

src/audio/level_multiplier/Kconfig

@@ -0,0 +1,12 @@
+# SPDX-License-Identifier: BSD-3-Clause
+
+config COMP_LEVEL_MULTIPLIER
+	tristate "Level multiplier component"
+	default y
+	help
+	  Select for Level multiplier component. This component
+          applies a fixed gain to audio. The amount of gain
+          is configured in the bytes control that is typically
+          set in boot time from topology. It can e.g. increase
+          capture sensitivity of voice applications by 20 dB
+          compared to media capture.

src/audio/level_multiplier/level_multiplier-generic.c

@@ -0,0 +1,266 @@
+// SPDX-License-Identifier: BSD-3-Clause
+//
+// Copyright(c) 2025 Intel Corporation.
+
+#include <sof/audio/module_adapter/module/generic.h>
+#include <sof/audio/component.h>
+#include <sof/audio/sink_api.h>
+#include <sof/audio/sink_source_utils.h>
+#include <sof/audio/source_api.h>
+#include <stdint.h>
+#include "level_multiplier.h"
+
+#define LEVEL_MULTIPLIER_S16_SHIFT	Q_SHIFT_BITS_32(15, LEVEL_MULTIPLIER_QXY_Y, 15)
+#define LEVEL_MULTIPLIER_S24_SHIFT	Q_SHIFT_BITS_64(23, LEVEL_MULTIPLIER_QXY_Y, 23)
+#define LEVEL_MULTIPLIER_S32_SHIFT	Q_SHIFT_BITS_64(31, LEVEL_MULTIPLIER_QXY_Y, 31)
+
+#if SOF_USE_HIFI(NONE, VOLUME)
+
+#if CONFIG_FORMAT_S16LE
+/**
+ * level_multiplier_s16() - Process S16_LE format.
+ * @mod: Pointer to module data.
+ * @source: Source for PCM samples data.
+ * @sink: Sink for PCM samples data.
+ * @frames: Number of audio data frames to process.
+ *
+ * This is the processing function for 16-bit signed integer PCM formats. The
+ * audio samples are copied from source to sink with gain defined in cd->gain.
+ *
+ * Return: Value zero for success, otherwise an error code.
+ */
+static int level_multiplier_s16(const struct processing_module *mod,
+				struct sof_source *source,
+				struct sof_sink *sink,
+				uint32_t frames)
+{
+	struct level_multiplier_comp_data *cd = module_get_private_data(mod);
+	const int32_t gain = cd->gain;
+	int16_t const *x, *x_start, *x_end;
+	int16_t *y, *y_start, *y_end;
+	int x_size, y_size;
+	int source_samples_without_wrap;
+	int samples_without_wrap;
+	int remaining_samples = frames * cd->channels;
+	int bytes = frames * cd->frame_bytes;
+	int ret;
+	int i;
+
+	ret = source_get_data_s16(source, bytes, &x, &x_start, &x_size);
+	if (ret)
+		return ret;
+
+	/* Similarly get pointer to sink data in circular buffer, buffer start and size. */
+	ret = sink_get_buffer_s16(sink, bytes, &y, &y_start, &y_size);
+	if (ret)
+		return ret;
+
+	/* Set helper pointers to buffer end for wrap check. Then loop until all
+	 * samples are processed.
+	 */
+	x_end = x_start + x_size;
+	y_end = y_start + y_size;
+	while (remaining_samples) {
+		/* Find out samples to process before first wrap or end of data. */
+		source_samples_without_wrap = x_end - x;
+		samples_without_wrap = y_end - y;
+		samples_without_wrap = MIN(samples_without_wrap, source_samples_without_wrap);
+		samples_without_wrap = MIN(samples_without_wrap, remaining_samples);
+		for (i = 0; i < samples_without_wrap; i++) {
+			*y = q_multsr_sat_32x32_16(*x, gain, LEVEL_MULTIPLIER_S16_SHIFT);
+			x++;
+			y++;
+		}
+
+		/* One of the buffers needs a wrap (or end of data), so check for wrap */
+		x = (x >= x_end) ? x - x_size : x;
+		y = (y >= y_end) ? y - y_size : y;
+
+		remaining_samples -= samples_without_wrap;
+	}
+
+	/* Update the source and sink for bytes consumed and produced. Return success. */
+	source_release_data(source, bytes);
+	sink_commit_buffer(sink, bytes);
+	return 0;
+}
+#endif /* CONFIG_FORMAT_S16LE */
+
+#if CONFIG_FORMAT_S24LE
+/**
+ * level_multiplier_s24() - Process S24_4LE format.
+ * @mod: Pointer to module data.
+ * @source: Source for PCM samples data.
+ * @sink: Sink for PCM samples data.
+ * @frames: Number of audio data frames to process.
+ *
+ * This is the processing function for 24-bit signed integer PCM formats. The
+ * audio samples are copied from source to sink with gain defined in cd->gain.
+ *
+ * Return: Value zero for success, otherwise an error code.
+ */
+static int level_multiplier_s24(const struct processing_module *mod,
+				struct sof_source *source,
+				struct sof_sink *sink,
+				uint32_t frames)
+{
+	struct level_multiplier_comp_data *cd = module_get_private_data(mod);
+	const int32_t gain = cd->gain;
+	int32_t const *x, *x_start, *x_end;
+	int32_t *y, *y_start, *y_end;
+	int x_size, y_size;
+	int source_samples_without_wrap;
+	int samples_without_wrap;
+	int remaining_samples = frames * cd->channels;
+	int bytes = frames * cd->frame_bytes;
+	int ret;
+	int i;
+
+	ret = source_get_data_s32(source, bytes, &x, &x_start, &x_size);
+	if (ret)
+		return ret;
+
+	/* Similarly get pointer to sink data in circular buffer, buffer start and size. */
+	ret = sink_get_buffer_s32(sink, bytes, &y, &y_start, &y_size);
+	if (ret)
+		return ret;
+
+	/* Set helper pointers to buffer end for wrap check. Then loop until all
+	 * samples are processed.
+	 */
+	x_end = x_start + x_size;
+	y_end = y_start + y_size;
+	while (remaining_samples) {
+		/* Find out samples to process before first wrap or end of data. */
+		source_samples_without_wrap = x_end - x;
+		samples_without_wrap = y_end - y;
+		samples_without_wrap = MIN(samples_without_wrap, source_samples_without_wrap);
+		samples_without_wrap = MIN(samples_without_wrap, remaining_samples);
+		for (i = 0; i < samples_without_wrap; i++) {
+			*y = q_multsr_sat_32x32_24(sign_extend_s24(*x), gain,
+						   LEVEL_MULTIPLIER_S24_SHIFT);
+			x++;
+			y++;
+		}
+
+		/* One of the buffers needs a wrap (or end of data), so check for wrap */
+		x = (x >= x_end) ? x - x_size : x;
+		y = (y >= y_end) ? y - y_size : y;
+
+		remaining_samples -= samples_without_wrap;
+	}
+
+	/* Update the source and sink for bytes consumed and produced. Return success. */
+	source_release_data(source, bytes);
+	sink_commit_buffer(sink, bytes);
+	return 0;
+}
+#endif /* CONFIG_FORMAT_S24LE */
+
+#if CONFIG_FORMAT_S32LE
+/**
+ * level_multiplier_s32() - Process S32_LE format.
+ * @mod: Pointer to module data.
+ * @source: Source for PCM samples data.
+ * @sink: Sink for PCM samples data.
+ * @frames: Number of audio data frames to process.
+ *
+ * This is the processing function for 32-bit signed integer PCM formats. The
+ * audio samples are copied from source to sink with gain defined in cd->gain.
+ *
+ * Return: Value zero for success, otherwise an error code.
+ */
+static int level_multiplier_s32(const struct processing_module *mod,
+				struct sof_source *source,
+				struct sof_sink *sink,
+				uint32_t frames)
+{
+	struct level_multiplier_comp_data *cd = module_get_private_data(mod);
+	const int32_t gain = cd->gain;
+	int32_t const *x, *x_start, *x_end;
+	int32_t *y, *y_start, *y_end;
+	int x_size, y_size;
+	int source_samples_without_wrap;
+	int samples_without_wrap;
+	int remaining_samples = frames * cd->channels;
+	int bytes = frames * cd->frame_bytes;
+	int ret;
+	int i;
+
+	ret = source_get_data_s32(source, bytes, &x, &x_start, &x_size);
+	if (ret)
+		return ret;
+
+	/* Similarly get pointer to sink data in circular buffer, buffer start and size. */
+	ret = sink_get_buffer_s32(sink, bytes, &y, &y_start, &y_size);
+	if (ret)
+		return ret;
+
+	/* Set helper pointers to buffer end for wrap check. Then loop until all
+	 * samples are processed.
+	 */
+	x_end = x_start + x_size;
+	y_end = y_start + y_size;
+	while (remaining_samples) {
+		/* Find out samples to process before first wrap or end of data. */
+		source_samples_without_wrap = x_end - x;
+		samples_without_wrap = y_end - y;
+		samples_without_wrap = MIN(samples_without_wrap, source_samples_without_wrap);
+		samples_without_wrap = MIN(samples_without_wrap, remaining_samples);
+		for (i = 0; i < samples_without_wrap; i++) {
+			*y = q_multsr_sat_32x32(*x, gain, LEVEL_MULTIPLIER_S32_SHIFT);
+			x++;
+			y++;
+		}
+
+		/* One of the buffers needs a wrap (or end of data), so check for wrap */
+		x = (x >= x_end) ? x - x_size : x;
+		y = (y >= y_end) ? y - y_size : y;
+
+		remaining_samples -= samples_without_wrap;
+	}
+
+	/* Update the source and sink for bytes consumed and produced. Return success. */
+	source_release_data(source, bytes);
+	sink_commit_buffer(sink, bytes);
+	return 0;
+}
+#endif /* CONFIG_FORMAT_S32LE */
+
+/* This struct array defines the used processing functions for
+ * the PCM formats
+ */
+const struct level_multiplier_proc_fnmap level_multiplier_proc_fnmap[] = {
+#if CONFIG_FORMAT_S16LE
+	{ SOF_IPC_FRAME_S16_LE, level_multiplier_s16 },
+#endif
+#if CONFIG_FORMAT_S24LE
+	{ SOF_IPC_FRAME_S24_4LE, level_multiplier_s24 },
+#endif
+#if CONFIG_FORMAT_S32LE
+	{ SOF_IPC_FRAME_S32_LE, level_multiplier_s32 },
+#endif
+};
+
+/**
+ * level_multiplier_find_proc_func() - Find suitable processing function.
+ * @src_fmt: Enum value for PCM format.
+ *
+ * This function finds the suitable processing function to use for
+ * the used PCM format. If not found, return NULL.
+ *
+ * Return: Pointer to processing function for the requested PCM format.
+ */
+level_multiplier_func level_multiplier_find_proc_func(enum sof_ipc_frame src_fmt)
+{
+	int i;
+
+	/* Find suitable processing function from map */
+	for (i = 0; i < ARRAY_SIZE(level_multiplier_proc_fnmap); i++)
+		if (src_fmt == level_multiplier_proc_fnmap[i].frame_fmt)
+			return level_multiplier_proc_fnmap[i].level_multiplier_proc_func;
+
+	return NULL;
+}
+
+#endif /* SOF_USE_HIFI(NONE, VOLUME) */