Chromium Code Reviews| 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 35e2d2c4d8c4e289105e9455f860932b83ad5ef4..21ff96218cb9d127e5ad84213f226468a591f81a 100644 |
| --- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
| +++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
| @@ -14,13 +14,16 @@ |
| #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/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" |
| namespace webrtc { |
| namespace test { |
| @@ -80,7 +83,11 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) { |
| AudioProcessingSimulator::AudioProcessingSimulator( |
| const SimulationSettings& settings) |
| - : settings_(settings), worker_queue_("file_writer_task_queue") { |
| + : settings_(settings), |
| + 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 +112,29 @@ 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 sumulated and an AEC dump is used as input, set |
|
peah-webrtc
2017/08/18 04:27:00
typo: sumulated
AleBzk
2017/08/18 07:49:46
Done.
|
| + // 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_.set_undo_mic_level(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( |
|
peah-webrtc
2017/08/18 04:27:00
As discussed offline, set_stream_analog_level must
AleBzk
2017/08/18 07:49:46
Done.
|
| + fake_recording_device_.mic_level())); |
| + } |
| + |
| + // Process the current audio frame. |
| if (fixed_interface) { |
| { |
| const auto st = ScopedTimer(mutable_proc_time()); |
| @@ -118,6 +148,12 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) { |
| out_config_, out_buf_->channels())); |
| } |
| + if (settings_.simulate_mic_gain) { |
| + // Store the mic level suggested by AGC. |
| + fake_recording_device_.set_mic_level( |
| + ap_->gain_control()->stream_analog_level()); |
| + } |
|
AleBzk
2017/07/26 13:42:31
As you pointed out, fake_recording_device should o
|
| + |
| if (buffer_writer_) { |
| buffer_writer_->Write(*out_buf_); |
| } |
| @@ -195,6 +231,8 @@ 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; |