Remove unused audio options and corresponding media constraints

- experimental AGC (aka googAutoGainControl2) removed in [1]
- experimental NS (aka googNoiseSuppression2) removed in [2]
- typing noise detection (aka googTypingNoiseDetection)
  removed in [3]
- cricket::AudioOptions::tx_agc_ are unused

[1] https://webrtc-review.googlesource.com/c/src/+/219463
[2] https://webrtc-review.googlesource.com/c/src/+/232128
[3] https://chromium-review.googlesource.com/c/chromium/src/+/1617352

Bug: webrtc:11226
Change-Id: Id1ecef3d3e193c210fc11832e16db4f84d866d14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35987}

api/audio_options.cc

@@ -55,13 +55,7 @@ void AudioOptions::SetAll(const AudioOptions& change) {
   SetFrom(&audio_jitter_buffer_enable_rtx_handling,
           change.audio_jitter_buffer_enable_rtx_handling);
   SetFrom(&typing_detection, change.typing_detection);
-  SetFrom(&experimental_agc, change.experimental_agc);
-  SetFrom(&experimental_ns, change.experimental_ns);
   SetFrom(&residual_echo_detector, change.residual_echo_detector);
-  SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
-  SetFrom(&tx_agc_digital_compression_gain,
-          change.tx_agc_digital_compression_gain);
-  SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
   SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
   SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
   SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
@@ -85,12 +79,7 @@ bool AudioOptions::operator==(const AudioOptions& o) const {
          audio_jitter_buffer_enable_rtx_handling ==
              o.audio_jitter_buffer_enable_rtx_handling &&
          typing_detection == o.typing_detection &&
-         experimental_agc == o.experimental_agc &&
-         experimental_ns == o.experimental_ns &&
          residual_echo_detector == o.residual_echo_detector &&
-         tx_agc_target_dbov == o.tx_agc_target_dbov &&
-         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
-         tx_agc_limiter == o.tx_agc_limiter &&
          combined_audio_video_bwe == o.combined_audio_video_bwe &&
          audio_network_adaptor == o.audio_network_adaptor &&
          audio_network_adaptor_config == o.audio_network_adaptor_config &&
@@ -119,13 +108,7 @@ std::string AudioOptions::ToString() const {
   ToStringIfSet(&result, "audio_jitter_buffer_enable_rtx_handling",
                 audio_jitter_buffer_enable_rtx_handling);
   ToStringIfSet(&result, "typing", typing_detection);
-  ToStringIfSet(&result, "experimental_agc", experimental_agc);
-  ToStringIfSet(&result, "experimental_ns", experimental_ns);
   ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector);
-  ToStringIfSet(&result, "tx_agc_target_dbov", tx_agc_target_dbov);
-  ToStringIfSet(&result, "tx_agc_digital_compression_gain",
-                tx_agc_digital_compression_gain);
-  ToStringIfSet(&result, "tx_agc_limiter", tx_agc_limiter);
   ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
   ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
   ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);

api/audio_options.h

@@ -64,16 +64,10 @@ struct RTC_EXPORT AudioOptions {
   // TODO(bugs.webrtc.org/11226): Remove.
   // Audio processing to detect typing.
   absl::optional<bool> typing_detection;
-  absl::optional<bool> experimental_agc;
-  absl::optional<bool> experimental_ns;
   // TODO(bugs.webrtc.org/11539): Deprecated, replaced by
   // webrtc::CreateEchoDetector() and injection when creating the audio
   // processing module.
   absl::optional<bool> residual_echo_detector;
-  // Note that tx_agc_* only applies to non-experimental AGC.
-  absl::optional<uint16_t> tx_agc_target_dbov;
-  absl::optional<uint16_t> tx_agc_digital_compression_gain;
-  absl::optional<bool> tx_agc_limiter;
   // Enable combined audio+bandwidth BWE.
   // TODO(pthatcher): This flag is set from the
   // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,

media/engine/webrtc_voice_engine.cc

@@ -404,14 +404,12 @@ void WebRtcVoiceEngine::Init() {
     options.noise_suppression = true;
     options.typing_detection = true;
 #endif
-    options.experimental_ns = false;
     options.highpass_filter = true;
     options.stereo_swapping = false;
     options.audio_jitter_buffer_max_packets = 200;
     options.audio_jitter_buffer_fast_accelerate = false;
     options.audio_jitter_buffer_min_delay_ms = 0;
     options.audio_jitter_buffer_enable_rtx_handling = false;
-    options.experimental_agc = false;
     bool error = ApplyOptions(options);
     RTC_DCHECK(error);
   }
@@ -464,17 +462,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
 // Override noise suppression options for Android.
 #if defined(WEBRTC_ANDROID)
   options.typing_detection = false;
-  options.experimental_ns = false;
 #endif
 
 // Set and adjust gain control options.
 #if defined(WEBRTC_IOS)
   // On iOS, VPIO provides built-in AGC.
   options.auto_gain_control = false;
-  options.experimental_agc = false;
   RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead.";
 #elif defined(WEBRTC_ANDROID)
-  options.experimental_agc = false;
 #endif
 
 #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
@@ -582,18 +577,8 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     return true;
   }
 
-  if (options.experimental_ns) {
-    experimental_ns_ = options.experimental_ns;
-  }
-
   webrtc::AudioProcessing::Config apm_config = ap->GetConfig();
 
-#if !(defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS))
-  if (experimental_ns_.has_value()) {
-    apm_config.transient_suppression.enabled = experimental_ns_.value();
-  }
-#endif
-
   if (options.echo_cancellation) {
     apm_config.echo_canceller.enabled = *options.echo_cancellation;
     apm_config.echo_canceller.mobile_mode = use_mobile_software_aec;
@@ -610,16 +595,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
         apm_config.gain_controller1.kAdaptiveAnalog;
 #endif
   }
-  if (options.tx_agc_target_dbov) {
-    apm_config.gain_controller1.target_level_dbfs = *options.tx_agc_target_dbov;
-  }
-  if (options.tx_agc_digital_compression_gain) {
-    apm_config.gain_controller1.compression_gain_db =
-        *options.tx_agc_digital_compression_gain;
-  }
-  if (options.tx_agc_limiter) {
-    apm_config.gain_controller1.enable_limiter = *options.tx_agc_limiter;
-  }
 
   if (options.highpass_filter) {
     apm_config.high_pass_filter.enabled = *options.highpass_filter;

media/engine/webrtc_voice_engine.h

@@ -124,9 +124,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   bool is_dumping_aec_ = false;
   bool initialized_ = false;
 
-  // Cache experimental_ns and apply in case they are missing in the audio
-  // options.
-  absl::optional<bool> experimental_ns_;
   // Jitter buffer settings for new streams.
   size_t audio_jitter_buffer_max_packets_ = 200;
   bool audio_jitter_buffer_fast_accelerate_ = false;

media/engine/webrtc_voice_engine_unittest.cc

@@ -2474,58 +2474,6 @@ TEST_P(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
   EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
 }
 
-TEST_P(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
-  EXPECT_TRUE(SetupSendStream());
-  EXPECT_CALL(*adm_, BuiltInAGCIsAvailable())
-      .Times(::testing::AtLeast(1))
-      .WillRepeatedly(Return(false));
-
-  if (!use_null_apm_) {
-    // Ensure default options.
-    VerifyGainControlEnabledCorrectly();
-    VerifyGainControlDefaultSettings();
-  }
-
-  const auto& agc_config = apm_config_.gain_controller1;
-
-  send_parameters_.options.auto_gain_control = false;
-  SetSendParameters(send_parameters_);
-  if (!use_null_apm_) {
-    EXPECT_FALSE(agc_config.enabled);
-  }
-  send_parameters_.options.auto_gain_control = absl::nullopt;
-
-  send_parameters_.options.tx_agc_target_dbov = 5;
-  SetSendParameters(send_parameters_);
-  if (!use_null_apm_) {
-    EXPECT_EQ(5, agc_config.target_level_dbfs);
-  }
-  send_parameters_.options.tx_agc_target_dbov = absl::nullopt;
-
-  send_parameters_.options.tx_agc_digital_compression_gain = 10;
-  SetSendParameters(send_parameters_);
-  if (!use_null_apm_) {
-    EXPECT_EQ(10, agc_config.compression_gain_db);
-  }
-  send_parameters_.options.tx_agc_digital_compression_gain = absl::nullopt;
-
-  send_parameters_.options.tx_agc_limiter = false;
-  SetSendParameters(send_parameters_);
-  if (!use_null_apm_) {
-    EXPECT_FALSE(agc_config.enable_limiter);
-  }
-  send_parameters_.options.tx_agc_limiter = absl::nullopt;
-
-  SetSendParameters(send_parameters_);
-  if (!use_null_apm_) {
-    // Expect all options to have been preserved.
-    EXPECT_FALSE(agc_config.enabled);
-    EXPECT_EQ(5, agc_config.target_level_dbfs);
-    EXPECT_EQ(10, agc_config.compression_gain_db);
-    EXPECT_FALSE(agc_config.enable_limiter);
-  }
-}
-
 TEST_P(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) {
   EXPECT_TRUE(SetupSendStream());
   send_parameters_.options.audio_network_adaptor = true;

sdk/media_constraints.cc

@@ -95,14 +95,8 @@ const char MediaConstraints::kValueFalse[] = "false";
 // Audio constraints.
 const char MediaConstraints::kGoogEchoCancellation[] = "googEchoCancellation";
 const char MediaConstraints::kAutoGainControl[] = "googAutoGainControl";
-const char MediaConstraints::kExperimentalAutoGainControl[] =
-    "googAutoGainControl2";
 const char MediaConstraints::kNoiseSuppression[] = "googNoiseSuppression";
-const char MediaConstraints::kExperimentalNoiseSuppression[] =
-    "googNoiseSuppression2";
 const char MediaConstraints::kHighpassFilter[] = "googHighpassFilter";
-const char MediaConstraints::kTypingNoiseDetection[] =
-    "googTypingNoiseDetection";
 const char MediaConstraints::kAudioMirroring[] = "googAudioMirroring";
 const char MediaConstraints::kAudioNetworkAdaptorConfig[] =
     "googAudioNetworkAdaptorConfig";
@@ -190,19 +184,10 @@ void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,
                              &options->echo_cancellation);
   ConstraintToOptional<bool>(constraints, MediaConstraints::kAutoGainControl,
                              &options->auto_gain_control);
-  ConstraintToOptional<bool>(constraints,
-                             MediaConstraints::kExperimentalAutoGainControl,
-                             &options->experimental_agc);
   ConstraintToOptional<bool>(constraints, MediaConstraints::kNoiseSuppression,
                              &options->noise_suppression);
-  ConstraintToOptional<bool>(constraints,
-                             MediaConstraints::kExperimentalNoiseSuppression,
-                             &options->experimental_ns);
   ConstraintToOptional<bool>(constraints, MediaConstraints::kHighpassFilter,
                              &options->highpass_filter);
-  ConstraintToOptional<bool>(constraints,
-                             MediaConstraints::kTypingNoiseDetection,
-                             &options->typing_detection);
   ConstraintToOptional<bool>(constraints, MediaConstraints::kAudioMirroring,
                              &options->stereo_swapping);
   ConstraintToOptional<std::string>(

sdk/media_constraints.h

@@ -59,12 +59,9 @@ class MediaConstraints {
   // These keys are google specific.
   static const char kGoogEchoCancellation[];  // googEchoCancellation
 
-  static const char kAutoGainControl[];               // googAutoGainControl
-  static const char kExperimentalAutoGainControl[];   // googAutoGainControl2
-  static const char kNoiseSuppression[];              // googNoiseSuppression
-  static const char kExperimentalNoiseSuppression[];  // googNoiseSuppression2
-  static const char kHighpassFilter[];                // googHighpassFilter
-  static const char kTypingNoiseDetection[];  // googTypingNoiseDetection
+  static const char kAutoGainControl[];       // googAutoGainControl
+  static const char kNoiseSuppression[];      // googNoiseSuppression
+  static const char kHighpassFilter[];        // googHighpassFilter
   static const char kAudioMirroring[];        // googAudioMirroring
   static const char
       kAudioNetworkAdaptorConfig[];  // googAudioNetworkAdaptorConfig