Enables robust delay validation in AEC delay logging.
* Explicitly disabled robust validation in AECM.
* Updated audio_processing_unittests for using robust delay validation in AEC.
* Updated output_data_float.pb (not needed for Android nor fixed point, since AECM is untouched).
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5368 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/aec/aec_core.c b/webrtc/modules/audio_processing/aec/aec_core.c
index bfa087c..4f4ce03 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/webrtc/modules/audio_processing/aec/aec_core.c
@@ -475,6 +475,17 @@
aec->extended_filter_enabled = 0;
aec->num_partitions = kNormalNumPartitions;
+ // Update the delay estimator with filter length. We use half the
+ // |num_partitions| to take the echo path into account. In practice we say
+ // that the echo has a duration of maximum half |num_partitions|, which is not
+ // true, but serves as a crude measure.
+ WebRtc_set_allowed_offset(aec->delay_estimator, aec->num_partitions / 2);
+ // TODO(bjornv): I currently hard coded the enable. Once we've established
+ // that AECM has no performance regression, robust_validation will be enabled
+ // all the time and the APIs to turn it on/off will be removed. Hence, remove
+ // this line then.
+ WebRtc_enable_robust_validation(aec->delay_estimator, 1);
+
// Default target suppression mode.
aec->nlp_mode = 1;
@@ -771,6 +782,8 @@
void WebRtcAec_enable_delay_correction(AecCore* self, int enable) {
self->extended_filter_enabled = enable;
self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions;
+ // Update the delay estimator with filter length. See InitAEC() for details.
+ WebRtc_set_allowed_offset(self->delay_estimator, self->num_partitions / 2);
}
int WebRtcAec_delay_correction_enabled(AecCore* self) {
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index fc94f1b..0f34874 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -266,6 +266,9 @@
aecm = NULL;
return -1;
}
+ // TODO(bjornv): Explicitly disable robust delay validation until no
+ // performance regression has been established. Then remove the line.
+ WebRtc_enable_robust_validation(aecm->delay_estimator, 0);
aecm->real_fft = WebRtcSpl_CreateRealFFT(PART_LEN_SHIFT);
if (aecm->real_fft == NULL) {
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 6eab1e8..6f1ee27 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -50,6 +50,8 @@
using webrtc::VoiceDetection;
namespace {
+// TODO(bjornv): This is not feasible until the functionality has been
+// re-implemented; see comment at the bottom of this file.
// When false, this will compare the output data with the results stored to
// file. This is the typical case. When the file should be updated, it can
// be set to true with the command-line switch --write_ref_data.
@@ -469,10 +471,10 @@
frame_queue.push(frame);
frame_delay--;
}
- // Run for 4.5 seconds, skipping statistics from the first second. We need
- // enough frames with audio to have reliable estimates, but as few as possible
- // to keep processing time down. 4.5 seconds seemed to be a good compromise
- // for this recording.
+ // Run for 4.5 seconds, skipping statistics from the first 2.5 seconds. We
+ // need enough frames with audio to have reliable estimates, but as few as
+ // possible to keep processing time down. 4.5 seconds seemed to be a good
+ // compromise for this recording.
for (int frame_count = 0; frame_count < 450; ++frame_count) {
webrtc::AudioFrame* frame = new AudioFrame();
frame->CopyFrom(tmp_frame);
@@ -496,7 +498,7 @@
frame_queue.pop();
delete frame;
- if (frame_count == 100) {
+ if (frame_count == 250) {
int median;
int std;
// Discard the first delay metrics to avoid convergence effects.
