Index: webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
index d2c274f460010af7976fc2a2f1032c8bb72ecc8a..eff15cda1d7c3ffaacc72275f0121ae8dc64b7f2 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
@@ -14,17 +14,23 @@ |
#include <iostream> |
#include <sstream> |
#include <string> |
+#include <utility> |
#include <vector> |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/logging.h" |
#include "webrtc/base/stringutils.h" |
#include "webrtc/common_audio/include/audio_util.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h" |
namespace webrtc { |
namespace test { |
namespace { |
+constexpr FakeRecordingDevice::DeviceKind kDefaultFakeRecDeviceKind = |
+ FakeRecordingDevice::DeviceKind::IDENTITY; |
+ |
void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) { |
RTC_CHECK_EQ(src.num_channels_, dest->num_channels()); |
RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames()); |
@@ -78,7 +84,13 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) { |
AudioProcessingSimulator::AudioProcessingSimulator( |
const SimulationSettings& settings) |
- : settings_(settings) { |
+ : settings_(settings), |
+ fake_recording_device_(FakeRecordingDevice::GetFakeRecDevice( |
+ settings_.simulate_mic_gain ? static_cast< |
+ FakeRecordingDevice::DeviceKind>(*settings.simulated_mic_kind) |
+ : kDefaultFakeRecDeviceKind, |
+ settings.initial_mic_level)) { |
+ RTC_DCHECK(fake_recording_device_); |
if (settings_.ed_graph_output_filename && |
settings_.ed_graph_output_filename->size() > 0) { |
residual_echo_likelihood_graph_writer_.open( |
@@ -103,6 +115,22 @@ AudioProcessingSimulator::ScopedTimer::~ScopedTimer() { |
} |
void AudioProcessingSimulator::ProcessStream(bool fixed_interface) { |
+ // Optionally use the fake recording device to simulate analog gain. |
+ RTC_DCHECK(fake_recording_device_); |
+ if (settings_.simulate_mic_gain) { |
+ if (fixed_interface) { |
+ fake_recording_device_->SimulateAnalogGain(&fwd_frame_); |
+ } else { |
+ fake_recording_device_->SimulateAnalogGain(in_buf_.get()); |
+ } |
+ } |
+ |
+ // Notify the current mic level to AGC. |
+ RTC_CHECK_EQ(AudioProcessing::kNoError, |
+ ap_->gain_control()->set_stream_analog_level( |
+ fake_recording_device_->mic_level())); |
+ |
+ // Process the current audio frame. |
if (fixed_interface) { |
{ |
const auto st = ScopedTimer(mutable_proc_time()); |
@@ -116,6 +144,10 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) { |
out_config_, out_buf_->channels())); |
} |
+ // Store the mic level suggested by AGC if required. |
+ fake_recording_device_->set_mic_level( |
+ ap_->gain_control()->stream_analog_level()); |
+ |
if (buffer_writer_) { |
buffer_writer_->Write(*out_buf_); |
} |
@@ -193,6 +225,8 @@ void AudioProcessingSimulator::SetupBuffersConfigsOutputs( |
rev_frame_.num_channels_ = reverse_input_num_channels; |
if (settings_.use_verbose_logging) { |
+ rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE); |
+ |
std::cout << "Sample rates:" << std::endl; |
std::cout << " Forward input: " << input_sample_rate_hz << std::endl; |
std::cout << " Forward output: " << output_sample_rate_hz << std::endl; |