Support platform tuning of WebRTC processing in ContentRendererClient

This allows platforms with known OS and hardware capabilities to apply
better tuned audio processing configuration.
The config is to parsed, validated and applied in
MediaStreamAudioProcessor.

The Config is passed as a std::string to keep ContentRendererClient
free from WebRTC dependencies, and make it easy for clients to store
the config. Another possible solution is to collect all related WebRTC
audio settings in a struct and have the function return or accept a
(forward-declared) pointer to said struct.

Bug: 892611
Binary-Size: Added dependency is needed to implement new feature. Binary size is very close to bot warn limit (20 kB vs 16 kB). Compensating for size increase seems possible, although not until next release.
Change-Id: Ib478c75ab42318b83ca4ef900587a964b0ceacbd
Reviewed-on: https://chromium-review.googlesource.com/c/1261437
Commit-Queue: Sam Zackrisson <saza@chromium.org>
Reviewed-by: Dmitry Gozman <dgozman@chromium.org>
Reviewed-by: Max Morin <maxmorin@chromium.org>
Cr-Commit-Position: refs/heads/master@{#598719}

content/public/renderer/content_renderer_client.cc

@@ -236,6 +236,11 @@ bool ContentRendererClient::ShouldEnforceWebRTCRoutingPreferences() {
   return true;
 }
 
+base::Optional<std::string>
+ContentRendererClient::WebRTCPlatformSpecificAudioProcessingConfiguration() {
+  return base::Optional<std::string>();
+}
+
 GURL ContentRendererClient::OverrideFlashEmbedWithHTML(const GURL& url) {
   return GURL();
 }

content/public/renderer/content_renderer_client.h

@@ -15,6 +15,7 @@
 #include "base/callback_forward.h"
 #include "base/files/file_path.h"
 #include "base/memory/ref_counted.h"
+#include "base/optional.h"
 #include "base/strings/string16.h"
 #include "base/task/task_scheduler/task_scheduler.h"
 #include "build/build_config.h"
@@ -378,6 +379,14 @@ class CONTENT_EXPORT ContentRendererClient {
   // routing logic, i.e. allowing multiple routes and non-proxied UDP.
   virtual bool ShouldEnforceWebRTCRoutingPreferences();
 
+  // Provides a default configuration of WebRTC audio processing, in JSON format
+  // with fields corresponding to webrtc::AudioProcessing::Config. Allows for a
+  // more functional tuning on platforms with known implementation and hardware
+  // limitations.
+  // This is currently not supported when running the Chrome audio service.
+  virtual base::Optional<std::string>
+  WebRTCPlatformSpecificAudioProcessingConfiguration();
+
   // Notifies that a worker context has been created. This function is called
   // from the worker thread.
   virtual void DidInitializeWorkerContextOnWorkerThread(

content/renderer/BUILD.gn

@@ -711,6 +711,8 @@ target(link_target_type, "renderer") {
     "//third_party/webrtc/api:libjingle_logging_api",
     "//third_party/webrtc/api:libjingle_peerconnection_api",
     "//third_party/webrtc/api:rtc_stats_api",
+    "//third_party/webrtc/api/audio:aec3_config",
+    "//third_party/webrtc/api/audio:aec3_config_json",
     "//third_party/webrtc/api/audio:aec3_factory",
     "//third_party/webrtc/api/audio_codecs:audio_codecs_api",
     "//third_party/webrtc/api/audio_codecs/L16:audio_decoder_L16",

content/renderer/media/stream/media_stream_audio_processor.cc

@@ -22,7 +22,9 @@
 #include "base/threading/thread_task_runner_handle.h"
 #include "base/trace_event/trace_event.h"
 #include "build/build_config.h"
+#include "content/public/common/content_client.h"
 #include "content/public/common/content_features.h"
+#include "content/public/renderer/content_renderer_client.h"
 #include "content/renderer/media/webrtc/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
 #include "media/base/audio_fifo.h"
@@ -30,6 +32,8 @@
 #include "media/base/channel_layout.h"
 #include "media/webrtc/echo_information.h"
 #include "media/webrtc/webrtc_switches.h"
+#include "third_party/webrtc/api/audio/echo_canceller3_config.h"
+#include "third_party/webrtc/api/audio/echo_canceller3_config_json.h"
 #include "third_party/webrtc/api/audio/echo_canceller3_factory.h"
 #include "third_party/webrtc/api/mediaconstraintsinterface.h"
 #include "third_party/webrtc/modules/audio_processing/include/audio_processing_statistics.h"
@@ -598,11 +602,22 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule(
   if (properties.echo_cancellation_type ==
       EchoCancellationType::kEchoCancellationAec3) {
     webrtc::EchoCanceller3Config aec3_config;
-    aec3_config.ep_strength.bounded_erl =
+    base::Optional<std::string> audio_processing_platform_config_json =
+        GetContentClient()
+            ->renderer()
+            ->WebRTCPlatformSpecificAudioProcessingConfiguration();
+    if (audio_processing_platform_config_json) {
+      aec3_config = webrtc::Aec3ConfigFromJsonString(
+          *audio_processing_platform_config_json);
+      bool config_parameters_already_valid =
+          webrtc::EchoCanceller3Config::Validate(&aec3_config);
+      RTC_DCHECK(config_parameters_already_valid);
+    }
+    aec3_config.ep_strength.bounded_erl |=
         base::FeatureList::IsEnabled(features::kWebRtcAecBoundedErlSetup);
-    aec3_config.echo_removal_control.has_clock_drift =
+    aec3_config.echo_removal_control.has_clock_drift |=
         base::FeatureList::IsEnabled(features::kWebRtcAecClockDriftSetup);
-    aec3_config.echo_audibility.use_stationary_properties =
+    aec3_config.echo_audibility.use_stationary_properties |=
         base::FeatureList::IsEnabled(features::kWebRtcAecNoiseTransparency);
 
     ap_builder.SetEchoControlFactory(