APM move: Implementing the APM for use in the audio service
I've placed it in media/webrtc as that made the most sense. I considered
making it its own taget (i.e. //media/webrtc:audio_processor) but it was
a bit cumbersome and other parts of media don't seem to do that.
This change includes moving some flags from content:: to media:: since
they need to be accessible by both audio processor implementations.
The new AudioProcessor code is similar to MediaStreamAudioProcessor, but
I've tried to simplify it as much as possible. It seems like the decisions
made for the old implementation were made under different circumstances
than we have now. The old implementation will run whenever the "APM in
Audio Service" feature isn't active. This'll ensure we're not changing
any behavior outside of the experiment.
The AudioProcessor is protected by a build flag to make sure it isn't built
on platforms where we don't have a sandboxed audio service.
For an outline of the project this CL is part of, see: https://docs.google.com/document/d/1u4POff_ts_1LE3WDLA_wDDFnUswdlsuHL5DsiTE0a3U/edit?usp=sharing
It's accessible to everyone @chromium.org.
Bug: 851959
Cq-Include-Trybots: luci.chromium.try:android_optional_gpu_tests_rel;luci.chromium.try:linux_optional_gpu_tests_rel;luci.chromium.try:mac_optional_gpu_tests_rel;luci.chromium.try:win_optional_gpu_tests_rel
Change-Id: Iec4cd2dd983b484921d5d0701d4a08bca21793b4
Reviewed-on: https://chromium-review.googlesource.com/1169463
Commit-Queue: Max Morin <[email protected]>
Reviewed-by: Dale Curtis <[email protected]>
Reviewed-by: John Abd-El-Malek <[email protected]>
Cr-Commit-Position: refs/heads/master@{#587050}diff --git a/content/browser/BUILD.gn b/content/browser/BUILD.gn
index bcf11a72..935d1008 100644
--- a/content/browser/BUILD.gn
+++ b/content/browser/BUILD.gn
@@ -109,6 +109,7 @@
"//media/mojo/interfaces",
"//media/mojo/interfaces:constants",
"//media/mojo/services",
+ "//media/webrtc",
"//mojo/core/embedder",
"//mojo/public/cpp/bindings",
"//mojo/public/cpp/platform",
diff --git a/content/browser/renderer_host/render_process_host_impl.cc b/content/browser/renderer_host/render_process_host_impl.cc
index 765b657..125c8cf 100644
--- a/content/browser/renderer_host/render_process_host_impl.cc
+++ b/content/browser/renderer_host/render_process_host_impl.cc
@@ -188,6 +188,7 @@
#include "media/base/media_switches.h"
#include "media/media_buildflags.h"
#include "media/mojo/services/video_decode_perf_history.h"
+#include "media/webrtc/webrtc_switches.h"
#include "mojo/public/cpp/bindings/associated_interface_ptr.h"
#include "mojo/public/cpp/bindings/strong_binding.h"
#include "mojo/public/cpp/system/platform_handle.h"
diff --git a/content/browser/utility_process_host.cc b/content/browser/utility_process_host.cc
index 9d7c979..042eccd 100644
--- a/content/browser/utility_process_host.cc
+++ b/content/browser/utility_process_host.cc
@@ -30,6 +30,7 @@
#include "content/public/common/service_manager_connection.h"
#include "content/public/common/service_names.mojom.h"
#include "media/base/media_switches.h"
+#include "media/webrtc/webrtc_switches.h"
#include "services/network/public/cpp/network_switches.h"
#include "services/service_manager/embedder/switches.h"
#include "services/service_manager/public/cpp/interface_provider.h"
@@ -313,6 +314,8 @@
switches::kFailAudioStreamCreation,
switches::kMuteAudio,
switches::kUseFileForFakeAudioCapture,
+ switches::kAecRefinedAdaptiveFilter,
+ switches::kAgcStartupMinVolume,
#if defined(OS_LINUX) || defined(OS_FREEBSD) || defined(OS_SOLARIS)
switches::kAlsaInputDevice,
switches::kAlsaOutputDevice,
diff --git a/content/public/common/content_features.cc b/content/public/common/content_features.cc
index e99a0b2..ac2b42d8 100644
--- a/content/public/common/content_features.cc
+++ b/content/public/common/content_features.cc
@@ -537,21 +537,6 @@
const base::Feature kWebPayments{"WebPayments",
base::FEATURE_ENABLED_BY_DEFAULT};
-// Informs the WebRTC Acoustic Echo Canceler (AEC) that echo path loss is
-// bounded.
-const base::Feature kWebRtcAecBoundedErlSetup{
- "WebRtcAecBoundedErlSetup", base::FEATURE_DISABLED_BY_DEFAULT};
-
-// Informs the WebRTC Acoustic Echo Canceler (AEC) that the playout and
-// capture is done using different clocks.
-const base::Feature kWebRtcAecClockDriftSetup{
- "WebRtcAecClockDriftSetup", base::FEATURE_DISABLED_BY_DEFAULT};
-
-// Informs the WebRTC Acoustic Echo Canceler (AEC) that the feature providing
-// echo canceller transparency to render noise should be used.
-const base::Feature kWebRtcAecNoiseTransparency{
- "WebRtcAecNoiseTransparency", base::FEATURE_DISABLED_BY_DEFAULT};
-
// Makes WebRTC use ECDSA certs by default (i.e., when no cert type was
// specified in JS).
const base::Feature kWebRtcEcdsaDefault{"WebRTC-EnableWebRtcEcdsa",
diff --git a/content/public/common/content_features.h b/content/public/common/content_features.h
index 71f3f7b1..19912c6 100644
--- a/content/public/common/content_features.h
+++ b/content/public/common/content_features.h
@@ -122,9 +122,6 @@
CONTENT_EXPORT extern const base::Feature kWebContentsOcclusion;
CONTENT_EXPORT extern const base::Feature kWebGLImageChromium;
CONTENT_EXPORT extern const base::Feature kWebPayments;
-CONTENT_EXPORT extern const base::Feature kWebRtcAecBoundedErlSetup;
-CONTENT_EXPORT extern const base::Feature kWebRtcAecClockDriftSetup;
-CONTENT_EXPORT extern const base::Feature kWebRtcAecNoiseTransparency;
CONTENT_EXPORT extern const base::Feature kWebRtcEcdsaDefault;
CONTENT_EXPORT extern const base::Feature kWebRtcHWH264Encoding;
CONTENT_EXPORT extern const base::Feature kWebRtcHWVP8Encoding;
diff --git a/content/public/common/content_switches.cc b/content/public/common/content_switches.cc
index f7e5123..049f139 100644
--- a/content/public/common/content_switches.cc
+++ b/content/public/common/content_switches.cc
@@ -13,21 +13,6 @@
// have an effect. 0 disables MSAA.
const char kAcceleratedCanvas2dMSAASampleCount[] = "canvas-msaa-sample-count";
-// Enables a new tuning of the WebRTC Acoustic Echo Canceler (AEC). The new
-// tuning aims at resolving two issues with the AEC:
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5777
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5778
-// TODO(hlundin): Remove this switch when experimentation is over;
-// crbug.com/603821.
-const char kAecRefinedAdaptiveFilter[] = "aec-refined-adaptive-filter";
-
-// Override the default minimum starting volume of the Automatic Gain Control
-// algorithm in WebRTC used with audio tracks from getUserMedia.
-// The valid range is 12-255. Values outside that range will be clamped
-// to the lowest or highest valid value inside WebRTC.
-// TODO(tommi): Remove this switch when crbug.com/555577 is fixed.
-const char kAgcStartupMinVolume[] = "agc-startup-min-volume";
-
// By default, file:// URIs cannot read other file:// URIs. This is an
// override for developers who need the old behavior for testing.
const char kAllowFileAccessFromFiles[] = "allow-file-access-from-files";
diff --git a/content/public/common/content_switches.h b/content/public/common/content_switches.h
index 4da6151b..081af901e 100644
--- a/content/public/common/content_switches.h
+++ b/content/public/common/content_switches.h
@@ -16,8 +16,6 @@
// All switches in alphabetical order. The switches should be documented
// alongside the definition of their values in the .cc file.
CONTENT_EXPORT extern const char kAcceleratedCanvas2dMSAASampleCount[];
-CONTENT_EXPORT extern const char kAecRefinedAdaptiveFilter[];
-CONTENT_EXPORT extern const char kAgcStartupMinVolume[];
CONTENT_EXPORT extern const char kAllowFileAccessFromFiles[];
CONTENT_EXPORT extern const char kAllowInsecureLocalhost[];
CONTENT_EXPORT extern const char kAllowLoopbackInPeerConnection[];
diff --git a/content/renderer/media/stream/media_stream_audio_processor.cc b/content/renderer/media/stream/media_stream_audio_processor.cc
index 4b37917..47f2533 100644
--- a/content/renderer/media/stream/media_stream_audio_processor.cc
+++ b/content/renderer/media/stream/media_stream_audio_processor.cc
@@ -23,13 +23,13 @@
#include "base/trace_event/trace_event.h"
#include "build/build_config.h"
#include "content/public/common/content_features.h"
-#include "content/public/common/content_switches.h"
#include "content/renderer/media/webrtc/webrtc_audio_device_impl.h"
#include "media/base/audio_converter.h"
#include "media/base/audio_fifo.h"
#include "media/base/audio_parameters.h"
#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_factory.h"
#include "third_party/webrtc/api/mediaconstraintsinterface.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing_statistics.h"
@@ -524,9 +524,9 @@
}
void MediaStreamAudioProcessor::GetStats(AudioProcessorStats* stats) {
- stats->typing_noise_detected =
- (base::subtle::Acquire_Load(&typing_detected_) != false);
- GetAudioProcessingStats(audio_processing_.get(), stats);
+ // This is the old GetStats interface from webrtc::AudioProcessorInterface.
+ // It should not be in use by Chrome any longer.
+ NOTREACHED();
}
webrtc::AudioProcessorInterface::AudioProcessorStatistics
