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Unified Diff: webrtc/modules/audio_processing/test/audio_processing_simulator.cc

Issue 2834643002: audioproc_f with simulated mic analog gain (Closed)
Patch Set: AEC dump + fake rec device bugfix Created 3 years, 3 months ago
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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 461fc71c5e16d272df8d7a23fb7d8e8eaba0a72f..9776d41745c65b55f661781ee0d55b4f9f5eb640 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -14,12 +14,15 @@
#include <iostream>
#include <sstream>
#include <string>
+#include <utility>
#include <vector>
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"
#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/stringutils.h"
namespace webrtc {
@@ -80,7 +83,12 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
AudioProcessingSimulator::AudioProcessingSimulator(
const SimulationSettings& settings)
- : settings_(settings), worker_queue_("file_writer_task_queue") {
+ : settings_(settings),
+ analog_mic_level_(settings.initial_mic_level),
+ fake_recording_device_(
+ settings.initial_mic_level,
+ settings_.simulate_mic_gain ? *settings.simulated_mic_kind : 0),
+ worker_queue_("file_writer_task_queue") {
if (settings_.ed_graph_output_filename &&
settings_.ed_graph_output_filename->size() > 0) {
residual_echo_likelihood_graph_writer_.open(
@@ -105,6 +113,43 @@ AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
}
void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
+ // Optionally use the fake recording device to simulate analog gain.
+ if (settings_.simulate_mic_gain) {
+ if (settings_.aec_dump_input_filename) {
+ // When the analog gain is simulated and an AEC dump is used as input, set
+ // the undo level to |aec_dump_mic_level_| to virtually restore the
+ // unmodified microphone signal level.
+ RTC_DCHECK(aec_dump_mic_level_);
+ fake_recording_device_.SetUndoMicLevel(aec_dump_mic_level_);
+ }
+
+ 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_.MicLevel()));
+ } else {
+ // Notify the current mic level to AGC.
+ if (settings_.aec_dump_input_filename) {
+ // AEC dump case.
+ RTC_DCHECK(aec_dump_mic_level_);
+ RTC_CHECK_EQ(
+ AudioProcessing::kNoError,
+ ap_->gain_control()->set_stream_analog_level(*aec_dump_mic_level_));
+ } else {
+ // Wav input file case.
+ RTC_CHECK_EQ(
+ AudioProcessing::kNoError,
+ ap_->gain_control()->set_stream_analog_level(analog_mic_level_));
peah-webrtc 2017/09/15 09:36:20 What about bundling line 148 with 143 using a cond
AleBzk 2017/09/22 12:33:55 Done.
+ }
+ }
+
+ // Process the current audio frame.
if (fixed_interface) {
{
const auto st = ScopedTimer(mutable_proc_time());
@@ -118,6 +163,14 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
out_config_, out_buf_->channels()));
}
+ // Store the mic level suggested by AGC.
+ // Note that when the analog gain is simulated and an AEC dump is used as
+ // input, |analog_mic_level_| will not be used with set_stream_analog_level().
+ analog_mic_level_ = ap_->gain_control()->stream_analog_level();
+ if (settings_.simulate_mic_gain) {
+ fake_recording_device_.SetMicLevel(analog_mic_level_);
+ }
+
if (buffer_writer_) {
buffer_writer_->Write(*out_buf_);
}
@@ -195,6 +248,8 @@ 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;

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