Remove usage of VoEAudioProcessing from WVoE/MC.

webrtc/media/engine/apm_helpers.cc

Index: webrtc/media/engine/apm_helpers.cc
diff --git a/webrtc/media/engine/apm_helpers.cc b/webrtc/media/engine/apm_helpers.cc
new file mode 100644
index 0000000000000000000000000000000000000000..8a0653fabea139eb0e0e139d8dfba2f8f3a6c122
--- /dev/null
+++ b/webrtc/media/engine/apm_helpers.cc
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/media/engine/apm_helpers.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"

[hlundin-webrtc, 2017/02/20 09:01:08]

Wrong order

[the sun, 2017/02/20 19:37:57]

Done.

+#include "webrtc/voice_engine/transmit_mixer.h"
+
+namespace webrtc {
+namespace apm_helpers {
+
+AgcConfig GetAgcConfig(AudioProcessing* apm) {
+  RTC_DCHECK(apm);
+  AgcConfig result;
+  result.targetLeveldBOv = apm->gain_control()->target_level_dbfs();
+  result.digitalCompressionGaindB = apm->gain_control()->compression_gain_db();
+  result.limiterEnable = apm->gain_control()->is_limiter_enabled();
+  return result;
+}
+
+void SetAgcConfig(AudioProcessing* apm,
+                  const AgcConfig& config) {
+  RTC_DCHECK(apm);
+  GainControl* gc = apm->gain_control();
+  if (gc->set_target_level_dbfs(config.targetLeveldBOv) != 0) {
+    LOG(LS_ERROR) << "Failed to set target level: " << config.targetLeveldBOv;
+  }
+  if (gc->set_compression_gain_db(config.digitalCompressionGaindB) != 0) {
+    LOG(LS_ERROR) << "Failed to set compression gain: "
+                  << config.digitalCompressionGaindB;
+  }
+  if (gc->enable_limiter(config.limiterEnable) != 0) {
+    LOG(LS_ERROR) << "Failed to set limiter on/off: " << config.limiterEnable;
+  }
+}
+
+void SetAgcStatus(AudioProcessing* apm,
+                  AudioDeviceModule* adm,
+                  bool enable,
+                  AgcModes mode) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(adm);
+  RTC_DCHECK(mode == kAgcFixedDigital || mode == kAgcAdaptiveAnalog);

[hlundin-webrtc, 2017/02/20 09:01:08]

This DCHECK is superfluous, since you are checking this in detail in the
#if-#else below.

[the sun, 2017/02/20 19:37:57]

Done.

+#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
+  RTC_DCHECK_EQ(kAgcFixedDigital, mode);
+  GainControl::Mode agc_mode = GainControl::kFixedDigital;
+#else
+  RTC_DCHECK_EQ(kAgcAdaptiveAnalog, mode);
+  GainControl::Mode agc_mode = GainControl::kAdaptiveAnalog;
+#endif
+  GainControl* gc = apm->gain_control();
+  if (gc->set_mode(agc_mode) != 0) {
+    LOG(LS_ERROR) << "Failed to set AGC mode: " << agc_mode;
+    return;
+  }
+  if (gc->Enable(enable) != 0) {
+    LOG(LS_ERROR) << "Failed to enable/disable AGC: " << enable;
+    return;
+  }
+  if (agc_mode != GainControl::kFixedDigital) {

[hlundin-webrtc, 2017/02/20 09:01:08]

Are we somehow depending on that the ADM AGC is already disabled when we select
kFixedDigital? Can we query it? DCHECK it?

[the sun, 2017/02/20 19:37:57]

Not sure what we depend on in reality - the SetAGC() impls on the Android/iOS
ADM are empty, except for check that we never try to enable AGC. I've updated
the comment and made this function always call SetAGC(). I think it is more
clear.

+    // Set Agc state in the ADM when adaptive Agc mode has been selected.
+    // Note that we also enable the ADM Agc when Adaptive Digital mode is
+    // used since we want to be able to provide the APM with updated mic
+    // levels when the user modifies the mic level manually.
+    if (adm->SetAGC(enable) != 0) {
+      LOG(LS_ERROR) << "Failed to set AGC mode in ADM: " << enable;
+      return;
+    }
+  }
+  LOG(LS_INFO) << "AGC set to " << enable << " with mode " << mode;
+}
+
+void SetEcStatus(AudioProcessing* apm,
+                 bool enable,
+                 EcModes mode) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(mode == kEcConference || mode == kEcAecm);

[hlundin-webrtc, 2017/02/20 09:01:08]

Consider RTC_DCHECK(...) << "mode: " << mode;
to get a printout of actual mode on failure.

[the sun, 2017/02/20 19:37:57]

Done.

+  EchoCancellation* ec = apm->echo_cancellation();
+  EchoControlMobile* ecm = apm->echo_control_mobile();
+  if (mode == kEcConference) {
+    // Disable the AECM before enabling the AEC.
+    if (enable && ecm->is_enabled() && ecm->Enable(false) != 0) {
+      LOG(LS_ERROR) << "Failed to disable AECM.";
+      return;
+    }
+    if (ec->Enable(enable) != 0) {
+      LOG(LS_ERROR) << "Failed to enable/disable AEC: " << enable;
+      return;
+    }
+    if (ec->set_suppression_level(EchoCancellation::kHighSuppression)
+        != 0) {
+      LOG(LS_ERROR) << "Failed to set high AEC aggressiveness.";
+      return;
+    }
+  } else {
+    // Disable the AEC before enabling the AECM.
+    if (enable && ec->is_enabled() && ec->Enable(false) != 0) {
+      LOG(LS_ERROR) << "Failed to disable AEC.";
+      return;
+    }
+    if (ecm->Enable(enable) != 0) {
+      LOG(LS_ERROR) << "Failed to enable/disable AECM: " << enable;
+      return;
+    }
+  }
+  LOG(LS_INFO) << "Echo control set to " << enable << " with mode " << mode;
+}
+
+void SetEcMetricsStatus(AudioProcessing* apm, bool enable) {
+  RTC_DCHECK(apm);
+  if ((apm->echo_cancellation()->enable_metrics(enable) != 0) ||
+      (apm->echo_cancellation()->enable_delay_logging(enable) != 0)) {
+    LOG(LS_ERROR) << "Failed to enable/disable EC metrics: " << enable;
+  }
+}
+
+void SetAecmMode(AudioProcessing* apm, bool enable_cng) {
+  RTC_DCHECK(apm);
+  EchoControlMobile* ecm = apm->echo_control_mobile();
+  RTC_DCHECK_EQ(EchoControlMobile::kSpeakerphone, ecm->routing_mode());
+  if (ecm->enable_comfort_noise(enable_cng) != 0) {
+      LOG(LS_ERROR) << "Failed to enable/disable CNG: " << enable_cng;

[hlundin-webrtc, 2017/02/20 09:01:07]

Alignment is off.

[the sun, 2017/02/20 19:37:57]

Done.

+    return;

[hlundin-webrtc, 2017/02/20 09:01:08]

Unnecessary return statement.

[the sun, 2017/02/20 19:37:58]

Added log statement in the success path instead.

+  }
+}
+
+void SetNsStatus(AudioProcessing* apm, bool enable) {
+  RTC_DCHECK(apm);
+  NoiseSuppression* ns = apm->noise_suppression();
+  if (ns->set_level(NoiseSuppression::kHigh) != 0) {
+    LOG(LS_ERROR) << "Failed to set high NS level.";
+    return;
+  }
+  if (ns->Enable(enable) != 0) {
+    LOG(LS_ERROR) << "Failed to enable/disable NS: " << enable;
+    return;
+  }
+  LOG(LS_INFO) << "NS set to " << enable;
+}
+
+void SetTypingDetectionStatus(AudioProcessing* apm, bool enable) {
+  RTC_DCHECK(apm);
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+  // Typing detection takes place in TransmitMixer::PrepareDemux() and
+  // TransmitMixer::TypingDetection(). The typing detection algorithm takes as
+  // input two booleans:
+  //   1. A signal whether a key was pressed during the audio frame.
+  //   2. Whether VAD is active or not.
+  // TransmitMixer will not even call the detector if APM has set kVadUnknown in
+  // the audio frame after near end processing, so enabling/disabling VAD is
+  // sufficient for turning typing detection on/off.
+  // TODO(solenberg): Rather than relying on a side effect, consider forcing the
+  //                  feature on/off in TransmitMixer.
+  VoiceDetection* vd = apm->voice_detection();
+  if (vd->Enable(enable)) {
+    LOG(LS_ERROR) << "Failed to enable/disable VAD: " << enable;
+    return;
+  }
+  if (vd->set_likelihood(VoiceDetection::kVeryLowLikelihood)) {
+    LOG(LS_ERROR) << "Failed to set low VAD likelihood.";
+    return;
+  }
+  LOG(LS_INFO) << "VAD set to " << enable;

[hlundin-webrtc, 2017/02/20 09:01:08]

Let the log text elaborate that VAD was set because of typing detection.

[the sun, 2017/02/20 19:37:57]

Done.

+#endif
+}
+}  // namespace apm_helpers
+}  // namespace webrtc