Reland: Drop the 16kHz sample rate restriction on AECM and zero out h…

…igher bands Landed originally here: https://codereview.webrtc.org/1774553002/ Revertede here: https://codereview.webrtc.org/1781893002/ TBR=solenberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1777093004 Cr-Commit-Position: refs/heads/master@{#12005}
GetStream · Mar 15, 2016 · 776593b · 776593b
1 parent f5d4786
commit 776593b
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Showing 6 changed files with 42 additions and 64 deletions.
diff --git a/data/audio_processing/output_data_fixed.pb b/data/audio_processing/output_data_fixed.pb
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -117,7 +117,6 @@ const size_t AudioProcessing::kNumNativeSampleRates =
     arraysize(AudioProcessing::kNativeSampleRatesHz);
 const int AudioProcessing::kMaxNativeSampleRateHz = AudioProcessing::
     kNativeSampleRatesHz[AudioProcessing::kNumNativeSampleRates - 1];
-const int AudioProcessing::kMaxAECMSampleRateHz = kSampleRate16kHz;
 
 AudioProcessing* AudioProcessing::Create() {
   Config config;
@@ -347,7 +346,7 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
 
   formats_.api_format = config;
 
-  // We process at the closest native rate >= min(input rate, output rate)...
+  // We process at the closest native rate >= min(input rate, output rate).
   const int min_proc_rate =
       std::min(formats_.api_format.input_stream().sample_rate_hz(),
                formats_.api_format.output_stream().sample_rate_hz());
@@ -358,11 +357,6 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
       break;
     }
   }
-  // ...with one exception.
-  if (public_submodules_->echo_control_mobile->is_enabled() &&
-      min_proc_rate > kMaxAECMSampleRateHz) {
-    fwd_proc_rate = kMaxAECMSampleRateHz;
-  }
 
   capture_nonlocked_.fwd_proc_format = StreamConfig(fwd_proc_rate);
 
@@ -595,12 +589,6 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
     return kBadSampleRateError;
   }
 
-  if (public_submodules_->echo_control_mobile->is_enabled() &&
-      frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
-    LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
-    return kUnsupportedComponentError;
-  }
-
   ProcessingConfig processing_config;
   {
     // Aquire lock for the access of api_format.
@@ -1254,7 +1242,9 @@ void AudioProcessingImpl::InitializeGainController() {
 
 void AudioProcessingImpl::InitializeEchoControlMobile() {
   public_submodules_->echo_control_mobile->Initialize(
-      proc_sample_rate_hz(), num_reverse_channels(), num_output_channels());
+      proc_split_sample_rate_hz(),
+      num_reverse_channels(),
+      num_output_channels());
 }
 
 void AudioProcessingImpl::InitializeLevelEstimator() {

diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -243,6 +243,12 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
 
       ++handle_index;
     }
+    for (size_t band = 1u; band < audio->num_bands(); ++band) {
+      memset(audio->split_bands(capture)[band],
+             0,
+             audio->num_frames_per_band() *
+                 sizeof(audio->split_bands(capture)[band][0]));
+    }
   }
   return AudioProcessing::kNoError;
 }

diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -508,7 +508,6 @@ class AudioProcessing {
   static const int kNativeSampleRatesHz[];
   static const size_t kNumNativeSampleRates;
   static const int kMaxNativeSampleRateHz;
-  static const int kMaxAECMSampleRateHz;
 
   static const int kChunkSizeMs = 10;
 };

diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -55,8 +55,8 @@ const google::protobuf::int32 kChannels[] = {1, 2};
 const int kSampleRates[] = {8000, 16000, 32000, 48000};
 
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
-// AECM doesn't support super-wb.
-const int kProcessSampleRates[] = {8000, 16000};
+// Android doesn't support 48kHz.
+const int kProcessSampleRates[] = {8000, 16000, 32000};
 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
 #endif
@@ -435,11 +435,7 @@ void ApmTest::SetUp() {
   frame_ = new AudioFrame();
   revframe_ = new AudioFrame();
 
-#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
-  Init(16000, 16000, 16000, 2, 2, 2, false);
-#else
   Init(32000, 32000, 32000, 2, 2, 2, false);
-#endif
 }
 
 void ApmTest::TearDown() {
@@ -1039,18 +1035,6 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
 }
 
 TEST_F(ApmTest, EchoControlMobile) {
-  // AECM won't use super-wideband.
-  SetFrameSampleRate(frame_, 32000);
-  EXPECT_NOERR(apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kBadSampleRateError,
-            apm_->echo_control_mobile()->Enable(true));
-  SetFrameSampleRate(frame_, 16000);
-  EXPECT_NOERR(apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_control_mobile()->Enable(true));
-  SetFrameSampleRate(frame_, 32000);
-  EXPECT_EQ(apm_->kUnsupportedComponentError, apm_->ProcessStream(frame_));
-
   // Turn AECM on (and AEC off)
   Init(16000, 16000, 16000, 2, 2, 2, false);
   EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
@@ -1974,6 +1958,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
                                         num_input_channels);
 
     int analog_level = 127;
+    size_t num_bad_chunks = 0;
     while (ReadFrame(far_file_, revframe_, revfloat_cb_.get()) &&
            ReadFrame(near_file_, frame_, float_cb_.get())) {
       frame_->vad_activity_ = AudioFrame::kVadUnknown;
@@ -2012,18 +1997,13 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
         float snr = ComputeSNR(output_int16.channels()[j],
                                output_cb.channels()[j],
                                samples_per_channel, &variance);
-  #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
-        // There are a few chunks in the fixed-point profile that give low SNR.
-        // Listening confirmed the difference is acceptable.
-        const float kVarianceThreshold = 150;
-        const float kSNRThreshold = 10;
-  #else
+
         const float kVarianceThreshold = 20;
         const float kSNRThreshold = 20;
-  #endif
+
         // Skip frames with low energy.
-        if (sqrt(variance) > kVarianceThreshold) {
-          EXPECT_LT(kSNRThreshold, snr);
+        if (sqrt(variance) > kVarianceThreshold && snr < kSNRThreshold) {
+          ++num_bad_chunks;
         }
       }
 
@@ -2039,6 +2019,16 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
       // Reset in case of downmixing.
       frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
     }
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+    const size_t kMaxNumBadChunks = 0;
+#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+    // There are a few chunks in the fixed-point profile that give low SNR.
+    // Listening confirmed the difference is acceptable.
+    const size_t kMaxNumBadChunks = 60;
+#endif
+    EXPECT_LE(num_bad_chunks, kMaxNumBadChunks);
+
     rewind(far_file_);
     rewind(near_file_);
   }
@@ -2560,9 +2550,9 @@ TEST_P(AudioProcessingTest, Formats) {
       } else {
         ref_rate = 8000;
       }
-#ifdef WEBRTC_AUDIOPROC_FIXED_PROFILE
+#ifdef WEBRTC_ARCH_ARM_FAMILY
       if (file_direction == kForward) {
-        ref_rate = std::min(ref_rate, 16000);
+        ref_rate = std::min(ref_rate, 32000);
       }
 #endif
       FILE* out_file = fopen(
@@ -2645,9 +2635,7 @@ TEST_P(AudioProcessingTest, Formats) {
         EXPECT_NE(0, expected_snr);
         std::cout << "SNR=" << snr << " dB" << std::endl;
       } else {
-        EXPECT_EQ(expected_snr, 0);
-        std::cout << "SNR="
-                  << "inf dB" << std::endl;
+        std::cout << "SNR=inf dB" << std::endl;
       }
 
       fclose(out_file);
@@ -2729,9 +2717,9 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(48000, 16000, 32000, 16000, 20, 20),
                     std::tr1::make_tuple(48000, 16000, 16000, 16000, 20, 0),
 
-                    std::tr1::make_tuple(44100, 48000, 48000, 48000, 20, 0),
-                    std::tr1::make_tuple(44100, 48000, 32000, 48000, 20, 30),
-                    std::tr1::make_tuple(44100, 48000, 16000, 48000, 20, 20),
+                    std::tr1::make_tuple(44100, 48000, 48000, 48000, 15, 0),
+                    std::tr1::make_tuple(44100, 48000, 32000, 48000, 15, 30),
+                    std::tr1::make_tuple(44100, 48000, 16000, 48000, 15, 20),
                     std::tr1::make_tuple(44100, 44100, 48000, 44100, 15, 20),
                     std::tr1::make_tuple(44100, 44100, 32000, 44100, 15, 15),
                     std::tr1::make_tuple(44100, 44100, 16000, 44100, 15, 15),
@@ -2742,15 +2730,15 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(44100, 16000, 32000, 16000, 20, 20),
                     std::tr1::make_tuple(44100, 16000, 16000, 16000, 20, 0),
 
-                    std::tr1::make_tuple(32000, 48000, 48000, 48000, 20, 0),
-                    std::tr1::make_tuple(32000, 48000, 32000, 48000, 20, 30),
-                    std::tr1::make_tuple(32000, 48000, 16000, 48000, 20, 20),
-                    std::tr1::make_tuple(32000, 44100, 48000, 44100, 15, 20),
-                    std::tr1::make_tuple(32000, 44100, 32000, 44100, 15, 15),
-                    std::tr1::make_tuple(32000, 44100, 16000, 44100, 15, 15),
-                    std::tr1::make_tuple(32000, 32000, 48000, 32000, 20, 35),
-                    std::tr1::make_tuple(32000, 32000, 32000, 32000, 20, 0),
-                    std::tr1::make_tuple(32000, 32000, 16000, 32000, 20, 20),
+                    std::tr1::make_tuple(32000, 48000, 48000, 48000, 35, 0),
+                    std::tr1::make_tuple(32000, 48000, 32000, 48000, 65, 30),
+                    std::tr1::make_tuple(32000, 48000, 16000, 48000, 40, 20),
+                    std::tr1::make_tuple(32000, 44100, 48000, 44100, 20, 20),
+                    std::tr1::make_tuple(32000, 44100, 32000, 44100, 20, 15),
+                    std::tr1::make_tuple(32000, 44100, 16000, 44100, 20, 15),
+                    std::tr1::make_tuple(32000, 32000, 48000, 32000, 35, 35),
+                    std::tr1::make_tuple(32000, 32000, 32000, 32000, 0, 0),
+                    std::tr1::make_tuple(32000, 32000, 16000, 32000, 40, 20),
                     std::tr1::make_tuple(32000, 16000, 48000, 16000, 20, 20),
                     std::tr1::make_tuple(32000, 16000, 32000, 16000, 20, 20),
                     std::tr1::make_tuple(32000, 16000, 16000, 16000, 20, 0),

diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
@@ -1125,11 +1125,6 @@ void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
       break;
     }
   }
-  if (audioproc_->echo_control_mobile()->is_enabled()) {
-    // AECM only supports 8 and 16 kHz.
-    _audioFrame.sample_rate_hz_ = std::min(
-        _audioFrame.sample_rate_hz_, AudioProcessing::kMaxAECMSampleRateHz);
-  }
   _audioFrame.num_channels_ = std::min(num_channels, num_codec_channels);
   RemixAndResample(audio, samples_per_channel, num_channels, sample_rate_hz,
                    &resampler_, &_audioFrame);