audio hal: fix input buffer size in non real time mode

Fix regression introduced by commit 0e46adf2 in input buffer size
calculation for fast input in non realtime mode.

Bug: 35452939
Test: Verified Skype audio
Change-Id: I76170e8a0c38f42b1be74f045472325d5c93add9
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9a1771c..85216bc 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -3792,13 +3792,13 @@
 #endif
             in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
             if (!in->realtime) {
+                in->config = pcm_config_audio_capture;
                 frame_size = audio_stream_in_frame_size(&in->stream);
                 buffer_size = get_input_buffer_size(config->sample_rate,
                                                     config->format,
                                                     channel_count,
                                                    is_low_latency);
                 in->config.period_size = buffer_size / frame_size;
-                in->config = pcm_config_audio_capture;
                 in->config.rate = config->sample_rate;
                 in->af_period_multiplier = 1;
             } else {