diff --git a/src/audio/google/google_rtc_audio_processing.c b/src/audio/google/google_rtc_audio_processing.c
index 2c41c2714370..b0952af4a217 100644
--- a/src/audio/google/google_rtc_audio_processing.c
+++ b/src/audio/google/google_rtc_audio_processing.c
@@ -602,6 +602,9 @@ static int google_rtc_audio_processing_prepare(struct processing_module *mod,
 		ret = -EINVAL;
 	}
 
+	cd->num_aec_reference_channels = MIN(ref_chan, REF_CHAN_MAX);
+	cd->num_capture_channels = MIN(mic_chan, MIC_CHAN_MAX);
+
 	if (ref_rate != mic_rate || ref_rate != out_rate ||
 	    ref_rate != CONFIG_COMP_GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ) {
 		comp_err(dev, "Incorrect source/sink sample rate, expect %d\n",
@@ -636,6 +639,11 @@ static int google_rtc_audio_processing_prepare(struct processing_module *mod,
 	cd->ref_frame_bytes = sizeof(mic_sample_t) * source_get_channels(sources[cd->aec_reference_source]);
 	cd->out_frame_bytes = cd->ref_frame_bytes;
 
+	ret = GoogleRtcAudioProcessingSetStreamFormats(cd->state, mic_rate,
+						       cd->num_capture_channels,
+						       cd->num_capture_channels,
+						       ref_rate, cd->num_aec_reference_channels);
+
 	/* Blobs sent during COMP_STATE_READY is assigned to blob_handler->data
 	 * directly, so comp_is_new_data_blob_available always returns false.
 	 */