diff --git a/src/audio/dai.c b/src/audio/dai.c
index fab6f36019ef..6a6b3386ef52 100644
--- a/src/audio/dai.c
+++ b/src/audio/dai.c
@@ -73,7 +73,10 @@ static void dai_dma_cb(void *arg, enum notify_id type, void *data)
 	struct comp_dev *dev = arg;
 	struct dai_data *dd = comp_get_drvdata(dev);
 	uint32_t bytes = next->elem.size;
+	struct comp_buffer *source;
+	struct comp_buffer *sink;
 	void *buffer_ptr;
+	int ret;
 
 	comp_dbg(dev, "dai_dma_cb()");
 
@@ -99,17 +102,32 @@ static void dai_dma_cb(void *arg, enum notify_id type, void *data)
 	}
 
 	if (dev->direction == SOF_IPC_STREAM_PLAYBACK) {
-		dma_buffer_copy_to(dd->local_buffer, dd->dma_buffer,
-				   dd->process, bytes);
+		ret = dma_buffer_copy_to(dd->local_buffer, dd->dma_buffer,
+					 dd->process, bytes);
 
 		buffer_ptr = dd->local_buffer->stream.r_ptr;
 	} else {
-		dma_buffer_copy_from(dd->dma_buffer, dd->local_buffer,
-				     dd->process, bytes);
+		ret = dma_buffer_copy_from(dd->dma_buffer, dd->local_buffer,
+					   dd->process, bytes);
 
 		buffer_ptr = dd->local_buffer->stream.w_ptr;
 	}
 
+	/* assert dma_buffer_copy succeed */
+	if (ret < 0) {
+		source = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
+					dd->local_buffer : dd->dma_buffer;
+		sink = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
+					dd->dma_buffer : dd->local_buffer;
+		comp_err(dev, "dai_dma_cb() dma buffer copy failed, dir %d bytes %d avail %d free %d",
+			 dev->direction, bytes,
+			 audio_stream_get_avail_samples(&source->stream) *
+				audio_stream_frame_bytes(&source->stream),
+			 audio_stream_get_free_samples(&sink->stream) *
+				audio_stream_frame_bytes(&sink->stream));
+		return;
+	}
+
 	/* update host position (in bytes offset) for drivers */
 	dev->position += bytes;
 	if (dd->dai_pos) {
diff --git a/src/audio/host.c b/src/audio/host.c
index 7051a1ccdbac..4a7f7c2b597a 100644
--- a/src/audio/host.c
+++ b/src/audio/host.c
@@ -130,13 +130,32 @@ static uint32_t host_dma_get_split(struct host_data *hd, uint32_t bytes)
 static void host_update_position(struct comp_dev *dev, uint32_t bytes)
 {
 	struct host_data *hd = comp_get_drvdata(dev);
+	struct comp_buffer *source;
+	struct comp_buffer *sink;
+	int ret;
+
 
 	if (dev->direction == SOF_IPC_STREAM_PLAYBACK)
-		dma_buffer_copy_from(hd->dma_buffer, hd->local_buffer,
-				     hd->process, bytes);
+		ret = dma_buffer_copy_from(hd->dma_buffer, hd->local_buffer,
+					   hd->process, bytes);
 	else
-		dma_buffer_copy_to(hd->local_buffer, hd->dma_buffer,
-				   hd->process, bytes);
+		ret = dma_buffer_copy_to(hd->local_buffer, hd->dma_buffer,
+					 hd->process, bytes);
+
+	/* assert dma_buffer_copy succeed */
+	if (ret < 0) {
+		source = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
+					hd->dma_buffer : hd->local_buffer;
+		sink = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
+					hd->local_buffer : hd->dma_buffer;
+		comp_err(dev, "host_update_position() dma buffer copy failed, dir %d bytes %d avail %d free %d",
+			 dev->direction, bytes,
+			 audio_stream_get_avail_samples(&source->stream) *
+				audio_stream_frame_bytes(&source->stream),
+			 audio_stream_get_free_samples(&sink->stream) *
+				audio_stream_frame_bytes(&sink->stream));
+		return;
+	}
 
 	dev->position += bytes;
 
diff --git a/src/audio/pcm_converter/pcm_converter.c b/src/audio/pcm_converter/pcm_converter.c
index 507ae460ff99..469f69cfa1d7 100644
--- a/src/audio/pcm_converter/pcm_converter.c
+++ b/src/audio/pcm_converter/pcm_converter.c
@@ -14,9 +14,9 @@
 #include <sof/audio/pcm_converter.h>
 #include <sof/debug/panic.h>
 
-void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
-			   struct audio_stream *sink, uint32_t ooffset,
-			   uint32_t samples, pcm_converter_lin_func converter)
+int pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
+			  struct audio_stream *sink, uint32_t ooffset,
+			  uint32_t samples, pcm_converter_lin_func converter)
 {
 	const int s_size_in = audio_stream_sample_bytes(source);
 	const int s_size_out = audio_stream_sample_bytes(sink);
@@ -30,8 +30,11 @@ void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
 	int chunk;
 	int N1, N2;
 
-	assert(audio_stream_get_avail_samples(source) >= samples + ioffset);
-	assert(audio_stream_get_free_samples(sink) >= samples + ooffset);
+	/* assert enough avail/free samples in source and sink buffer */
+	if (audio_stream_get_avail_samples(source) < samples + ioffset)
+		return -EINVAL;
+	if (audio_stream_get_free_samples(sink) < samples + ooffset)
+		return -EINVAL;
 
 	while (i < samples) {
 		/* calculate chunk size */
@@ -51,4 +54,6 @@ void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
 		w_ptr = audio_stream_wrap(sink, w_ptr + chunk * s_size_out);
 		i += chunk;
 	}
+
+	return samples;
 }
diff --git a/src/audio/pcm_converter/pcm_converter_generic.c b/src/audio/pcm_converter/pcm_converter_generic.c
index da77e26296f3..eb61aa697647 100644
--- a/src/audio/pcm_converter/pcm_converter_generic.c
+++ b/src/audio/pcm_converter/pcm_converter_generic.c
@@ -26,9 +26,9 @@
 
 #if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE
 
-static void pcm_convert_s16_to_s24(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s16_to_s24(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	uint32_t buff_frag = 0;
 	int16_t *src;
@@ -41,11 +41,13 @@ static void pcm_convert_s16_to_s24(const struct audio_stream *source,
 		*dst = *src << 8;
 		buff_frag++;
 	}
+
+	return samples;
 }
 
-static void pcm_convert_s24_to_s16(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s24_to_s16(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	uint32_t buff_frag = 0;
 	int32_t *src;
@@ -58,15 +60,17 @@ static void pcm_convert_s24_to_s16(const struct audio_stream *source,
 		*dst = sat_int16(Q_SHIFT_RND(sign_extend_s24(*src), 23, 15));
 		buff_frag++;
 	}
+
+	return samples;
 }
 
 #endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE */
 
 #if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE
 
-static void pcm_convert_s16_to_s32(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s16_to_s32(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	uint32_t buff_frag = 0;
 	int16_t *src;
@@ -79,11 +83,13 @@ static void pcm_convert_s16_to_s32(const struct audio_stream *source,
 		*dst = *src << 16;
 		buff_frag++;
 	}
+
+	return samples;
 }
 
-static void pcm_convert_s32_to_s16(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s32_to_s16(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	uint32_t buff_frag = 0;
 	int32_t *src;
@@ -96,15 +102,17 @@ static void pcm_convert_s32_to_s16(const struct audio_stream *source,
 		*dst = sat_int16(Q_SHIFT_RND(*src, 31, 15));
 		buff_frag++;
 	}
+
+	return samples;
 }
 
 #endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE */
 
 #if CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE
 
-static void pcm_convert_s24_to_s32(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s24_to_s32(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	uint32_t buff_frag = 0;
 	int32_t *src;
@@ -117,11 +125,13 @@ static void pcm_convert_s24_to_s32(const struct audio_stream *source,
 		*dst = *src << 8;
 		buff_frag++;
 	}
+
+	return samples;
 }
 
-static void pcm_convert_s32_to_s24(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s32_to_s24(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	uint32_t buff_frag = 0;
 	int32_t *src;
@@ -134,6 +144,8 @@ static void pcm_convert_s32_to_s24(const struct audio_stream *source,
 		*dst = sat_int24(Q_SHIFT_RND(*src, 31, 23));
 		buff_frag++;
 	}
+
+	return samples;
 }
 
 #endif /* CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE */
@@ -272,20 +284,20 @@ static void pcm_convert_f_to_s16_lin(const void *psrc, void *pdst,
 		dst[i] = sat_int16(_pcm_convert_f_to_i(src[i], 15));
 }
 
-static void pcm_convert_s16_to_f(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s16_to_f(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_s16_to_f_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_s16_to_f_lin);
 }
 
-static void pcm_convert_f_to_s16(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_f_to_s16(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_f_to_s16_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_f_to_s16_lin);
 }
 #endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S16LE */
 
@@ -316,20 +328,20 @@ static void pcm_convert_f_to_s24_lin(const void *psrc, void *pdst,
 		dst[i] = sat_int24(_pcm_convert_f_to_i(src[i], 23));
 }
 
-static void pcm_convert_s24_to_f(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s24_to_f(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_s24_to_f_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_s24_to_f_lin);
 }
 
-static void pcm_convert_f_to_s24(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_f_to_s24(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_f_to_s24_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_f_to_s24_lin);
 }
 #endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S24LE */
 
@@ -360,20 +372,20 @@ static void pcm_convert_f_to_s32_lin(const void *psrc, void *pdst,
 		dst[i] = _pcm_convert_f_to_i(src[i], 31);
 }
 
-static void pcm_convert_s32_to_f(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s32_to_f(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_s32_to_f_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_s32_to_f_lin);
 }
 
-static void pcm_convert_f_to_s32(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_f_to_s32(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_f_to_s32_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_f_to_s32_lin);
 }
 #endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S32LE */
 
diff --git a/src/audio/pcm_converter/pcm_converter_hifi3.c b/src/audio/pcm_converter/pcm_converter_hifi3.c
index 5f4f94632ba0..1ed000872531 100644
--- a/src/audio/pcm_converter/pcm_converter_hifi3.c
+++ b/src/audio/pcm_converter/pcm_converter_hifi3.c
@@ -45,9 +45,9 @@ static void pcm_converter_setup_circular(const struct audio_stream *source)
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
  */
-static void pcm_convert_s16_to_s24(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s16_to_s24(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	ae_int16 *in = audio_stream_read_frag(source, ioffset, sizeof(int16_t));
 	ae_int32 *out = audio_stream_write_frag(sink, ooffset, sizeof(int32_t));
@@ -59,7 +59,7 @@ static void pcm_convert_s16_to_s24(const struct audio_stream *source,
 
 	/* nothing to do */
 	if (!samples)
-		return;
+		return samples;
 
 	/* required alignment for AE_L16X4_XC */
 	while (!IS_ALIGNED((uintptr_t)in, 8)) {
@@ -77,7 +77,7 @@ static void pcm_convert_s16_to_s24(const struct audio_stream *source,
 			    sizeof(ae_int32));
 
 		if (++i == samples)
-			return;
+			return samples;
 	}
 
 	in16x4 = (ae_int16x4 *)in;
@@ -123,6 +123,8 @@ static void pcm_convert_s16_to_s24(const struct audio_stream *source,
 		AE_S32_L_XC(AE_SRAI32(AE_CVT32X2F16_32(sample), 8), out,
 			    sizeof(ae_int32));
 	}
+
+	return samples;
 }
 
 /**
@@ -146,10 +148,11 @@ static ae_int32x2 pcm_shift_s24_to_s16(ae_int32x2 sample)
  * \param[in,out] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s24_to_s16(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s24_to_s16(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
 						sizeof(int32_t));
@@ -164,7 +167,7 @@ static void pcm_convert_s24_to_s16(const struct audio_stream *source,
 
 	/* nothing to do */
 	if (!samples)
-		return;
+		return samples;
 
 	/* required alignment for AE_L32X2_XC */
 	if (!IS_ALIGNED((uintptr_t)in, 8)) {
@@ -236,7 +239,7 @@ static void pcm_convert_s24_to_s16(const struct audio_stream *source,
 
 		/* no more samples to process */
 		if (leftover == 1)
-			return;
+			return samples;
 
 		/* store one 16 bit sample */
 		AE_S16_0_XC(AE_MOVAD16_0(sample), (ae_int16 *)out,
@@ -244,6 +247,8 @@ static void pcm_convert_s24_to_s16(const struct audio_stream *source,
 
 		leftover -= 2;
 	}
+
+	return samples;
 }
 
 #endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE */
@@ -255,10 +260,11 @@ static void pcm_convert_s24_to_s16(const struct audio_stream *source,
  * \param[in,out] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s16_to_s32(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s16_to_s32(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	ae_int16 *in = audio_stream_read_frag(source, ioffset,
 					      sizeof(int16_t));
@@ -272,7 +278,7 @@ static void pcm_convert_s16_to_s32(const struct audio_stream *source,
 
 	/* nothing to do */
 	if (!samples)
-		return;
+		return samples;
 
 	/* required alignment for AE_L16X4_XC */
 	while (!IS_ALIGNED((uintptr_t)in, 8)) {
@@ -289,7 +295,7 @@ static void pcm_convert_s16_to_s32(const struct audio_stream *source,
 		AE_S32_L_XC(AE_CVT32X2F16_32(sample), out, sizeof(ae_int32));
 
 		if (++i == samples)
-			return;
+			return samples;
 	}
 
 	in16x4 = (ae_int16x4 *)in;
@@ -332,6 +338,8 @@ static void pcm_convert_s16_to_s32(const struct audio_stream *source,
 		/* store one 32 bit sample */
 		AE_S32_L_XC(AE_CVT32X2F16_32(sample), out, sizeof(ae_int32));
 	}
+
+	return samples;
 }
 
 /**
@@ -339,10 +347,11 @@ static void pcm_convert_s16_to_s32(const struct audio_stream *source,
  * \param[in,out] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s32_to_s16(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s32_to_s16(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
 						sizeof(int32_t));
@@ -357,7 +366,7 @@ static void pcm_convert_s32_to_s16(const struct audio_stream *source,
 
 	/* nothing to do */
 	if (!samples)
-		return;
+		return samples;
 
 	/* required alignment for AE_L32X2_XC */
 	if (!IS_ALIGNED((uintptr_t)in, 8)) {
@@ -425,7 +434,7 @@ static void pcm_convert_s32_to_s16(const struct audio_stream *source,
 
 		/* no more samples to process */
 		if (leftover == 1)
-			return;
+			return samples;
 
 		/* store one 16 bit sample */
 		AE_S16_0_XC(AE_MOVAD16_0(sample), (ae_int16 *)out,
@@ -433,6 +442,8 @@ static void pcm_convert_s32_to_s16(const struct audio_stream *source,
 
 		leftover -= 2;
 	}
+
+	return samples;
 }
 
 #endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE */
@@ -444,10 +455,11 @@ static void pcm_convert_s32_to_s16(const struct audio_stream *source,
  * \param[in,out] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s24_to_s32(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s24_to_s32(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
 						sizeof(int32_t));
@@ -459,7 +471,7 @@ static void pcm_convert_s24_to_s32(const struct audio_stream *source,
 
 	/* nothing to do */
 	if (!samples)
-		return;
+		return samples;
 
 	/* required alignment for AE_L32X2_XC */
 	if (!IS_ALIGNED((uintptr_t)in, 8)) {
@@ -499,7 +511,7 @@ static void pcm_convert_s24_to_s32(const struct audio_stream *source,
 
 	/* no more samples to process */
 	if (i == samples)
-		return;
+		return samples;
 
 	/* set source as circular buffer */
 	pcm_converter_setup_circular(source);
@@ -512,6 +524,8 @@ static void pcm_convert_s24_to_s32(const struct audio_stream *source,
 
 	/* shift left and store one 32 bit sample */
 	AE_S32_L_XC(AE_SLAI32(sample, 8), (ae_int32 *)out, 0);
+
+	return samples;
 }
 
 /**
@@ -534,10 +548,11 @@ static ae_int32x2 pcm_shift_s32_to_s24(ae_int32x2 sample)
  * \param[in,out] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s32_to_s24(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s32_to_s24(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples)
 {
 	ae_int32x2 *in = audio_stream_read_frag(source, ioffset,
 						sizeof(int32_t));
@@ -549,7 +564,7 @@ static void pcm_convert_s32_to_s24(const struct audio_stream *source,
 
 	/* nothing to do */
 	if (!samples)
-		return;
+		return samples;
 
 	/* required alignment for AE_L32X2_XC */
 	if (!IS_ALIGNED((uintptr_t)in, 8)) {
@@ -590,7 +605,7 @@ static void pcm_convert_s32_to_s24(const struct audio_stream *source,
 
 	/* no more samples to process */
 	if (i == samples)
-		return;
+		return samples;
 
 	/* set source as circular buffer */
 	pcm_converter_setup_circular(source);
@@ -604,6 +619,8 @@ static void pcm_convert_s32_to_s24(const struct audio_stream *source,
 	/* shift right and store one 32 bit sample */
 	sample = pcm_shift_s32_to_s24(sample);
 	AE_S32_L_XC(sample, (ae_int32 *)out, 0);
+
+	return samples;
 }
 
 #endif /* CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE */
@@ -675,13 +692,14 @@ static void pcm_convert_f_to_s16_lin(const void *psrc, void *pdst,
  * \param[in] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s16_to_f(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s16_to_f(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_s16_to_f_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_s16_to_f_lin);
 }
 
 /**
@@ -689,13 +707,14 @@ static void pcm_convert_s16_to_f(const struct audio_stream *source,
  * \param[in] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_f_to_s16(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_f_to_s16(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_f_to_s16_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_f_to_s16_lin);
 }
 #endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S16LE */
 
@@ -773,13 +792,14 @@ static void pcm_convert_f_to_s24_lin(const void *psrc, void *pdst,
  * \param[in] psrc source linear buffer.
  * \param[out] pdst destination linear buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s24_to_f(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s24_to_f(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_s24_to_f_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_s24_to_f_lin);
 }
 
 /**
@@ -787,13 +807,14 @@ static void pcm_convert_s24_to_f(const struct audio_stream *source,
  * \param[in] psrc source linear buffer.
  * \param[out] pdst destination linear buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_f_to_s24(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_f_to_s24(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_f_to_s24_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_f_to_s24_lin);
 }
 #endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_24LE */
 
@@ -868,13 +889,14 @@ static void pcm_convert_f_to_s32_lin(const void *psrc, void *pdst,
  * \param[in] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_s32_to_f(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_s32_to_f(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_s32_to_f_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_s32_to_f_lin);
 }
 
 /**
@@ -882,13 +904,14 @@ static void pcm_convert_s32_to_f(const struct audio_stream *source,
  * \param[in] source Source buffer.
  * \param[in,out] sink Destination buffer.
  * \param[in] samples Number of samples to process.
+ * \return error code or number of processed samples.
  */
-static void pcm_convert_f_to_s32(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t samples)
+static int pcm_convert_f_to_s32(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t samples)
 {
-	pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
-			      pcm_convert_f_to_s32_lin);
+	return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
+				     pcm_convert_f_to_s32_lin);
 }
 #endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_32LE */
 #endif /* XCHAL_HAVE_FP */
diff --git a/src/include/sof/audio/audio_stream.h b/src/include/sof/audio/audio_stream.h
index d46b5c3f264b..ed1c57717101 100644
--- a/src/include/sof/audio/audio_stream.h
+++ b/src/include/sof/audio/audio_stream.h
@@ -550,8 +550,9 @@ audio_stream_frames_without_wrap(const struct audio_stream *source,
  * @param sink Sink buffer.
  * @param ooffset Offset (in samples) in sink buffer to start writing to.
  * @param samples Number of samples to copy.
+ * @return number of processed samples.
  */
-static inline void audio_stream_copy(const struct audio_stream *source,
+static inline int audio_stream_copy(const struct audio_stream *source,
 				     uint32_t ioffset,
 				     struct audio_stream *sink,
 				     uint32_t ooffset, uint32_t samples)
@@ -582,6 +583,8 @@ static inline void audio_stream_copy(const struct audio_stream *source,
 		src = audio_stream_wrap(source, src);
 		snk = audio_stream_wrap(sink, snk);
 	}
+
+	return samples;
 }
 
 /** @}*/
diff --git a/src/include/sof/audio/pcm_converter.h b/src/include/sof/audio/pcm_converter.h
index 74e78450653c..50d0718d675a 100644
--- a/src/include/sof/audio/pcm_converter.h
+++ b/src/include/sof/audio/pcm_converter.h
@@ -41,10 +41,11 @@ struct audio_stream;
  * \param sink output buffer, write pointer is not modified
  * \param ooffset offset to first sample in sink stream
  * \param samples number of samples to convert
+ * \return error code or number of processed samples.
  */
-typedef void (*pcm_converter_func)(const struct audio_stream *source,
-				   uint32_t ioffset, struct audio_stream *sink,
-				   uint32_t ooffset, uint32_t samples);
+typedef int (*pcm_converter_func)(const struct audio_stream *source,
+				  uint32_t ioffset, struct audio_stream *sink,
+				  uint32_t ooffset, uint32_t samples);
 
 /**
  * \brief PCM conversion function interface for data in linear buffer
@@ -100,9 +101,10 @@ pcm_get_conversion_function(enum sof_ipc_frame in,
  * \param ooffset offset to first sample in sink stream
  * \param samples number of samples to convert
  * \param converter core conversion function working on linear memory regions
+ * \return error code or number of processed samples
  */
-void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
-			   struct audio_stream *sink, uint32_t ooffset,
-			   uint32_t samples, pcm_converter_lin_func converter);
+int pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
+			  struct audio_stream *sink, uint32_t ooffset,
+			  uint32_t samples, pcm_converter_lin_func converter);
 
 #endif /* __SOF_AUDIO_PCM_CONVERTER_H__ */
diff --git a/src/include/sof/lib/dma.h b/src/include/sof/lib/dma.h
index ef5383498d8b..59c3a398ea30 100644
--- a/src/include/sof/lib/dma.h
+++ b/src/include/sof/lib/dma.h
@@ -222,9 +222,9 @@ struct dma_info {
 };
 
 struct audio_stream;
-typedef void (*dma_process_func)(const struct audio_stream *source,
-				 uint32_t ioffset, struct audio_stream *sink,
-				 uint32_t ooffset, uint32_t frames);
+typedef int (*dma_process_func)(const struct audio_stream *source,
+				uint32_t ioffset, struct audio_stream *sink,
+				uint32_t ooffset, uint32_t frames);
 
 /**
  * \brief API to initialize a platform DMA controllers.
@@ -607,12 +607,12 @@ typedef void (*dma_process)(const struct audio_stream *,
 			    struct audio_stream *, uint32_t);
 
 /* copies data from DMA buffer using provided processing function */
-void dma_buffer_copy_from(struct comp_buffer *source, struct comp_buffer *sink,
-			  dma_process_func process, uint32_t source_bytes);
+int dma_buffer_copy_from(struct comp_buffer *source, struct comp_buffer *sink,
+			 dma_process_func process, uint32_t source_bytes);
 
 /* copies data to DMA buffer using provided processing function */
-void dma_buffer_copy_to(struct comp_buffer *source, struct comp_buffer *sink,
-			dma_process_func process, uint32_t sink_bytes);
+int dma_buffer_copy_to(struct comp_buffer *source, struct comp_buffer *sink,
+		       dma_process_func process, uint32_t sink_bytes);
 
 /* generic DMA DSP <-> Host copier */
 
diff --git a/src/lib/dma.c b/src/lib/dma.c
index 41d9293376be..9ae2450d8d8b 100644
--- a/src/lib/dma.c
+++ b/src/lib/dma.c
@@ -189,20 +189,21 @@ void dma_sg_free(struct dma_sg_elem_array *elem_array)
 	dma_sg_init(elem_array);
 }
 
-void dma_buffer_copy_from(struct comp_buffer *source, struct comp_buffer *sink,
-			  dma_process_func process, uint32_t source_bytes)
+int dma_buffer_copy_from(struct comp_buffer *source, struct comp_buffer *sink,
+			 dma_process_func process, uint32_t source_bytes)
 {
 	struct audio_stream *istream = &source->stream;
 	uint32_t samples = source_bytes /
 			   audio_stream_sample_bytes(istream);
 	uint32_t sink_bytes = audio_stream_sample_bytes(&sink->stream) *
 			      samples;
+	int ret;
 
 	/* source buffer contains data copied by DMA */
 	audio_stream_invalidate(istream, source_bytes);
 
 	/* process data */
-	process(istream, 0, &sink->stream, 0, samples);
+	ret = process(istream, 0, &sink->stream, 0, samples);
 
 	buffer_writeback(sink, sink_bytes);
 
@@ -212,21 +213,24 @@ void dma_buffer_copy_from(struct comp_buffer *source, struct comp_buffer *sink,
 	 */
 	audio_stream_consume(istream, source_bytes);
 	comp_update_buffer_produce(sink, sink_bytes);
+
+	return ret;
 }
 
-void dma_buffer_copy_to(struct comp_buffer *source, struct comp_buffer *sink,
-			dma_process_func process, uint32_t sink_bytes)
+int dma_buffer_copy_to(struct comp_buffer *source, struct comp_buffer *sink,
+		       dma_process_func process, uint32_t sink_bytes)
 {
 	struct audio_stream *ostream = &sink->stream;
 	uint32_t samples = sink_bytes /
 			   audio_stream_sample_bytes(ostream);
 	uint32_t source_bytes = audio_stream_sample_bytes(&source->stream) *
 			      samples;
+	int ret;
 
 	buffer_invalidate(source, source_bytes);
 
 	/* process data */
-	process(&source->stream, 0, ostream, 0, samples);
+	ret = process(&source->stream, 0, ostream, 0, samples);
 
 	/* sink buffer contains data meant to copied to DMA */
 	audio_stream_writeback(ostream, sink_bytes);
@@ -237,4 +241,6 @@ void dma_buffer_copy_to(struct comp_buffer *source, struct comp_buffer *sink,
 	 */
 	audio_stream_produce(ostream, sink_bytes);
 	comp_update_buffer_consume(source, source_bytes);
+
+	return ret;
 }