This CL is partially reverting the effects that

were added in https://codereview.webrtc.org/1773173002. The reason for the revert is that for some scenarios that CL causes problems in the coherence estimate used in the AEC, which in turn causes echo leakage. The reason for not reverting the actual CL is that it would cause subsequent CLs to be reverted as well. Therefore the choice was made to in this CK instead revert the effects of that CL. With the changes in this CL, the behavior is bitexact to what it was before the CL mentioned above. TBR=aluebs@webrtc.org BUG=webrtc:5725 Review URL: https://codereview.webrtc.org/1867483003 Cr-Commit-Position: refs/heads/master@{#12259}
GetStream · Apr 6, 2016 · 0bf612b · 0bf612b
1 parent fc3ef3e
commit 0bf612b
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Showing 5 changed files with 15 additions and 12 deletions.
diff --git a/data/audio_processing/output_data_fixed.pb b/data/audio_processing/output_data_fixed.pb
diff --git a/data/audio_processing/output_data_float.pb b/data/audio_processing/output_data_float.pb
diff --git a/data/audio_processing/output_data_mac.pb b/data/audio_processing/output_data_mac.pb
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -366,15 +366,18 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
       formats_.api_format.input_stream().sample_rate_hz(),
       formats_.api_format.output_stream().sample_rate_hz())));
 
-  int rev_proc_rate = ClosestNativeRate(std::min(
-      formats_.api_format.reverse_input_stream().sample_rate_hz(),
-      formats_.api_format.reverse_output_stream().sample_rate_hz()));
-  // If the forward sample rate is 8 kHz, the reverse stream is also processed
-  // at this rate.
+  // We normally process the reverse stream at 16 kHz. Unless...
+  int rev_proc_rate = kSampleRate16kHz;
   if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
+    // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    rev_proc_rate = std::max(rev_proc_rate, static_cast<int>(kSampleRate16kHz));
+    if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
+        kSampleRate32kHz) {
+      // ...or the input is at 32 kHz, in which case we use the splitting
+      // filter rather than the resampler.
+      rev_proc_rate = kSampleRate32kHz;
+    }
   }
 
   // Always downmix the reverse stream to mono for analysis. This has been
@@ -1148,11 +1151,11 @@ bool AudioProcessingImpl::is_rev_processed() const {
 
 bool AudioProcessingImpl::rev_synthesis_needed() const {
   return (is_rev_processed() &&
-          is_multi_band(formats_.rev_proc_format.sample_rate_hz()));
+          formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz);
 }
 
 bool AudioProcessingImpl::rev_analysis_needed() const {
-  return is_multi_band(formats_.rev_proc_format.sample_rate_hz()) &&
+  return formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
          (is_rev_processed() ||
           public_submodules_->echo_cancellation->is_enabled() ||
           public_submodules_->echo_control_mobile->is_enabled() ||

diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -2644,8 +2644,8 @@ INSTANTIATE_TEST_CASE_P(
     CommonFormats,
     AudioProcessingTest,
     testing::Values(std::tr1::make_tuple(48000, 48000, 48000, 48000, 0, 0),
-                    std::tr1::make_tuple(48000, 48000, 32000, 48000, 35, 30),
-                    std::tr1::make_tuple(48000, 48000, 16000, 48000, 35, 20),
+                    std::tr1::make_tuple(48000, 48000, 32000, 48000, 40, 30),
+                    std::tr1::make_tuple(48000, 48000, 16000, 48000, 40, 20),
                     std::tr1::make_tuple(48000, 44100, 48000, 44100, 20, 20),
                     std::tr1::make_tuple(48000, 44100, 32000, 44100, 20, 15),
                     std::tr1::make_tuple(48000, 44100, 16000, 44100, 20, 15),
@@ -2692,7 +2692,7 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
                     std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
                     std::tr1::make_tuple(16000, 16000, 48000, 16000, 40, 20),
-                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
+                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 50, 20),
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 
 #elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
@@ -2748,7 +2748,7 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
                     std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
                     std::tr1::make_tuple(16000, 16000, 48000, 16000, 35, 20),
-                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 35, 20),
+                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 #endif