From 135259ac8fac1fd88e948742e6ab473425a43d88 Mon Sep 17 00:00:00 2001 From: peah Date: Fri, 28 Oct 2016 03:12:11 -0700 Subject: [PATCH] In order to be able to analyze the AGC behavior on aecdump recordings in an efficient manner, it is important to be able to use a standardized analysis script. For this to be feasible, data log points should be present. This CL adds those logpoints as well as the framework needed to for those to work. BUG=webrtc:6564 Review-Url: https://codereview.webrtc.org/2457783003 Cr-Commit-Position: refs/heads/master@{#14812} --- .../audio_processing/audio_processing_impl.cc | 7 ++++--- .../gain_control_for_experimental_agc.cc | 20 +++++++++++++++++-- .../gain_control_for_experimental_agc.h | 13 +++++++++--- .../audio_processing/gain_control_impl.cc | 8 ++++++++ .../audio_processing/gain_control_impl.h | 4 ++++ 5 files changed, 44 insertions(+), 8 deletions(-) diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index 189d7095e0..b1a41934a0 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -435,9 +435,6 @@ int AudioProcessingImpl::InitializeLocked() { capture_audiobuffer_num_channels, formats_.api_format.output_stream().num_frames())); - public_submodules_->gain_control->Initialize(num_proc_channels(), - proc_sample_rate_hz()); - public_submodules_->echo_cancellation->Initialize( proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(), num_proc_channels()); @@ -450,6 +447,9 @@ int AudioProcessingImpl::InitializeLocked() { public_submodules_->echo_control_mobile->Initialize( proc_split_sample_rate_hz(), num_reverse_channels(), num_output_channels()); + + public_submodules_->gain_control->Initialize(num_proc_channels(), + proc_sample_rate_hz()); if (constants_.use_experimental_agc) { if (!private_submodules_->agc_manager.get()) { private_submodules_->agc_manager.reset(new AgcManagerDirect( @@ -460,6 +460,7 @@ int AudioProcessingImpl::InitializeLocked() { private_submodules_->agc_manager->Initialize(); private_submodules_->agc_manager->SetCaptureMuted( capture_.output_will_be_muted); + public_submodules_->gain_control_for_experimental_agc->Initialize(); } InitializeTransient(); InitializeBeamformer(); diff --git a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc index 2ef88c03aa..f8063865b6 100644 --- a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc +++ b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc @@ -13,15 +13,23 @@ #include "webrtc/base/checks.h" #include "webrtc/base/criticalsection.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" +#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" namespace webrtc { +int GainControlForExperimentalAgc::instance_counter_ = 0; + GainControlForExperimentalAgc::GainControlForExperimentalAgc( GainControl* gain_control, rtc::CriticalSection* crit_capture) - : real_gain_control_(gain_control), + : data_dumper_(new ApmDataDumper(instance_counter_)), + real_gain_control_(gain_control), volume_(0), - crit_capture_(crit_capture) {} + crit_capture_(crit_capture) { + instance_counter_++; +} + +GainControlForExperimentalAgc::~GainControlForExperimentalAgc() = default; int GainControlForExperimentalAgc::Enable(bool enable) { return real_gain_control_->Enable(enable); @@ -33,12 +41,16 @@ bool GainControlForExperimentalAgc::is_enabled() const { int GainControlForExperimentalAgc::set_stream_analog_level(int level) { rtc::CritScope cs_capture(crit_capture_); + data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level", 1, + &level); volume_ = level; return AudioProcessing::kNoError; } int GainControlForExperimentalAgc::stream_analog_level() { rtc::CritScope cs_capture(crit_capture_); + data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1, + &volume_); return volume_; } @@ -101,4 +113,8 @@ int GainControlForExperimentalAgc::GetMicVolume() { return volume_; } +void GainControlForExperimentalAgc::Initialize() { + data_dumper_->InitiateNewSetOfRecordings(); +} + } // namespace webrtc diff --git a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h index 4fbd05c685..75d87d17c3 100644 --- a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h +++ b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h @@ -19,6 +19,8 @@ namespace webrtc { +class ApmDataDumper; + // This class has two main purposes: // // 1) It is returned instead of the real GainControl after the new AGC has been @@ -33,8 +35,9 @@ namespace webrtc { class GainControlForExperimentalAgc : public GainControl, public VolumeCallbacks { public: - explicit GainControlForExperimentalAgc(GainControl* gain_control, - rtc::CriticalSection* crit_capture); + GainControlForExperimentalAgc(GainControl* gain_control, + rtc::CriticalSection* crit_capture); + ~GainControlForExperimentalAgc() override; // GainControl implementation. int Enable(bool enable) override; @@ -58,11 +61,15 @@ class GainControlForExperimentalAgc : public GainControl, void SetMicVolume(int volume) override; int GetMicVolume() override; + void Initialize(); + private: + std::unique_ptr data_dumper_; GainControl* real_gain_control_; int volume_; rtc::CriticalSection* crit_capture_; - RTC_DISALLOW_COPY_AND_ASSIGN(GainControlForExperimentalAgc); + static int instance_counter_; + RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlForExperimentalAgc); }; } // namespace webrtc diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc index 8f707fc3b5..81469dde01 100644 --- a/webrtc/modules/audio_processing/gain_control_impl.cc +++ b/webrtc/modules/audio_processing/gain_control_impl.cc @@ -14,6 +14,7 @@ #include "webrtc/base/optional.h" #include "webrtc/modules/audio_processing/audio_buffer.h" #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h" +#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" namespace webrtc { @@ -84,10 +85,13 @@ class GainControlImpl::GainController { RTC_DISALLOW_COPY_AND_ASSIGN(GainController); }; +int GainControlImpl::instance_counter_ = 0; + GainControlImpl::GainControlImpl(rtc::CriticalSection* crit_render, rtc::CriticalSection* crit_capture) : crit_render_(crit_render), crit_capture_(crit_capture), + data_dumper_(new ApmDataDumper(instance_counter_)), mode_(kAdaptiveAnalog), minimum_capture_level_(0), maximum_capture_level_(255), @@ -239,6 +243,7 @@ int GainControlImpl::compression_gain_db() const { // TODO(ajm): ensure this is called under kAdaptiveAnalog. int GainControlImpl::set_stream_analog_level(int level) { rtc::CritScope cs(crit_capture_); + data_dumper_->DumpRaw("gain_control_set_stream_analog_level", 1, &level); was_analog_level_set_ = true; if (level < minimum_capture_level_ || level > maximum_capture_level_) { @@ -251,6 +256,8 @@ int GainControlImpl::set_stream_analog_level(int level) { int GainControlImpl::stream_analog_level() { rtc::CritScope cs(crit_capture_); + data_dumper_->DumpRaw("gain_control_stream_analog_level", 1, + &analog_capture_level_); // TODO(ajm): enable this assertion? //RTC_DCHECK_EQ(kAdaptiveAnalog, mode_); @@ -385,6 +392,7 @@ bool GainControlImpl::is_limiter_enabled() const { void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) { rtc::CritScope cs_render(crit_render_); rtc::CritScope cs_capture(crit_capture_); + data_dumper_->InitiateNewSetOfRecordings(); num_proc_channels_ = rtc::Optional(num_proc_channels); sample_rate_hz_ = rtc::Optional(sample_rate_hz); diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h index 812b88cb8d..bd56ed452d 100644 --- a/webrtc/modules/audio_processing/gain_control_impl.h +++ b/webrtc/modules/audio_processing/gain_control_impl.h @@ -23,6 +23,7 @@ namespace webrtc { +class ApmDataDumper; class AudioBuffer; class GainControlImpl : public GainControl { @@ -69,6 +70,8 @@ class GainControlImpl : public GainControl { rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_); rtc::CriticalSection* const crit_capture_; + std::unique_ptr data_dumper_; + bool enabled_ = false; Mode mode_ GUARDED_BY(crit_capture_); @@ -86,6 +89,7 @@ class GainControlImpl : public GainControl { rtc::Optional num_proc_channels_ GUARDED_BY(crit_capture_); rtc::Optional sample_rate_hz_ GUARDED_BY(crit_capture_); + static int instance_counter_; RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl); }; } // namespace webrtc