| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <string.h> |
| |
| #include <iostream> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/test/aec_dump_based_simulator.h" |
| #include "modules/audio_processing/test/audio_processing_simulator.h" |
| #include "modules/audio_processing/test/audioproc_float_impl.h" |
| #include "modules/audio_processing/test/wav_based_simulator.h" |
| #include "rtc_base/flags.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| const int kParameterNotSpecifiedValue = -10000; |
| |
| const char kUsageDescription[] = |
| "Usage: audioproc_f [options] -i <input.wav>\n" |
| " or\n" |
| " audioproc_f [options] -dump_input <aec_dump>\n" |
| "\n\n" |
| "Command-line tool to simulate a call using the audio " |
| "processing module, either based on wav files or " |
| "protobuf debug dump recordings.\n"; |
| |
| WEBRTC_DEFINE_string(dump_input, "", "Aec dump input filename"); |
| WEBRTC_DEFINE_string(dump_output, "", "Aec dump output filename"); |
| WEBRTC_DEFINE_string(i, "", "Forward stream input wav filename"); |
| WEBRTC_DEFINE_string(o, "", "Forward stream output wav filename"); |
| WEBRTC_DEFINE_string(ri, "", "Reverse stream input wav filename"); |
| WEBRTC_DEFINE_string(ro, "", "Reverse stream output wav filename"); |
| WEBRTC_DEFINE_string(artificial_nearend, "", "Artificial nearend wav filename"); |
| WEBRTC_DEFINE_int(output_num_channels, |
| kParameterNotSpecifiedValue, |
| "Number of forward stream output channels"); |
| WEBRTC_DEFINE_int(reverse_output_num_channels, |
| kParameterNotSpecifiedValue, |
| "Number of Reverse stream output channels"); |
| WEBRTC_DEFINE_int(output_sample_rate_hz, |
| kParameterNotSpecifiedValue, |
| "Forward stream output sample rate in Hz"); |
| WEBRTC_DEFINE_int(reverse_output_sample_rate_hz, |
| kParameterNotSpecifiedValue, |
| "Reverse stream output sample rate in Hz"); |
| WEBRTC_DEFINE_bool(fixed_interface, |
| false, |
| "Use the fixed interface when operating on wav files"); |
| WEBRTC_DEFINE_int(aec, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the echo canceller"); |
| WEBRTC_DEFINE_int(aecm, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the mobile echo controller"); |
| WEBRTC_DEFINE_int(ed, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate (0) the residual echo detector"); |
| WEBRTC_DEFINE_string(ed_graph, |
| "", |
| "Output filename for graph of echo likelihood"); |
| WEBRTC_DEFINE_int(agc, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the AGC"); |
| WEBRTC_DEFINE_int(agc2, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the AGC2"); |
| WEBRTC_DEFINE_int(pre_amplifier, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the pre amplifier"); |
| WEBRTC_DEFINE_int(hpf, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the high-pass filter"); |
| WEBRTC_DEFINE_int(ns, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the noise suppressor"); |
| WEBRTC_DEFINE_int(ts, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the transient suppressor"); |
| WEBRTC_DEFINE_int(vad, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the voice activity detector"); |
| WEBRTC_DEFINE_int(le, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the level estimator"); |
| WEBRTC_DEFINE_bool( |
| all_default, |
| false, |
| "Activate all of the default components (will be overridden by any " |
| "other settings)"); |
| WEBRTC_DEFINE_int(aec_suppression_level, |
| kParameterNotSpecifiedValue, |
| "Set the aec suppression level (0-2)"); |
| WEBRTC_DEFINE_int(delay_agnostic, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the AEC delay agnostic mode"); |
| WEBRTC_DEFINE_int(extended_filter, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the AEC extended filter mode"); |
| WEBRTC_DEFINE_int( |
| aec3, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the experimental AEC mode AEC3"); |
| WEBRTC_DEFINE_int(experimental_agc, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the experimental AGC"); |
| WEBRTC_DEFINE_int( |
| experimental_agc_disable_digital_adaptive, |
| kParameterNotSpecifiedValue, |
| "Force-deactivate (1) digital adaptation in " |
| "experimental AGC. Digital adaptation is active by default (0)."); |
| WEBRTC_DEFINE_int(experimental_agc_analyze_before_aec, |
| kParameterNotSpecifiedValue, |
| "Make level estimation happen before AEC" |
| " in the experimental AGC. After AEC is the default (0)"); |
| WEBRTC_DEFINE_int( |
| experimental_agc_agc2_level_estimator, |
| kParameterNotSpecifiedValue, |
| "AGC2 level estimation" |
| " in the experimental AGC. AGC1 level estimation is the default (0)"); |
| WEBRTC_DEFINE_int( |
| refined_adaptive_filter, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the refined adaptive filter functionality"); |
| WEBRTC_DEFINE_int(agc_mode, |
| kParameterNotSpecifiedValue, |
| "Specify the AGC mode (0-2)"); |
| WEBRTC_DEFINE_int(agc_target_level, |
| kParameterNotSpecifiedValue, |
| "Specify the AGC target level (0-31)"); |
| WEBRTC_DEFINE_int(agc_limiter, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the level estimator"); |
| WEBRTC_DEFINE_int(agc_compression_gain, |
| kParameterNotSpecifiedValue, |
| "Specify the AGC compression gain (0-90)"); |
| WEBRTC_DEFINE_float(agc2_enable_adaptive_gain, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) the AGC2 adaptive gain"); |
| WEBRTC_DEFINE_float(agc2_fixed_gain_db, 0.f, "AGC2 fixed gain (dB) to apply"); |
| WEBRTC_DEFINE_float(pre_amplifier_gain_factor, |
| 1.f, |
| "Pre-amplifier gain factor (linear) to apply"); |
| WEBRTC_DEFINE_int(vad_likelihood, |
| kParameterNotSpecifiedValue, |
| "Specify the VAD likelihood (0-3)"); |
| WEBRTC_DEFINE_int(ns_level, |
| kParameterNotSpecifiedValue, |
| "Specify the NS level (0-3)"); |
| WEBRTC_DEFINE_int(stream_delay, |
| kParameterNotSpecifiedValue, |
| "Specify the stream delay in ms to use"); |
| WEBRTC_DEFINE_int(use_stream_delay, |
| kParameterNotSpecifiedValue, |
| "Activate (1) or deactivate(0) reporting the stream delay"); |
| WEBRTC_DEFINE_int(stream_drift_samples, |
| kParameterNotSpecifiedValue, |
| "Specify the number of stream drift samples to use"); |
| WEBRTC_DEFINE_int(initial_mic_level, 100, "Initial mic level (0-255)"); |
| WEBRTC_DEFINE_int( |
| simulate_mic_gain, |
| 0, |
| "Activate (1) or deactivate(0) the analog mic gain simulation"); |
| WEBRTC_DEFINE_int( |
| simulated_mic_kind, |
| kParameterNotSpecifiedValue, |
| "Specify which microphone kind to use for microphone simulation"); |
| WEBRTC_DEFINE_bool(performance_report, false, "Report the APM performance "); |
| WEBRTC_DEFINE_bool(verbose, false, "Produce verbose output"); |
| WEBRTC_DEFINE_bool(quiet, |
| false, |
| "Avoid producing information about the progress."); |
| WEBRTC_DEFINE_bool(bitexactness_report, |
| false, |
| "Report bitexactness for aec dump result reproduction"); |
| WEBRTC_DEFINE_bool(discard_settings_in_aecdump, |
| false, |
| "Discard any config settings specified in the aec dump"); |
| WEBRTC_DEFINE_bool(store_intermediate_output, |
| false, |
| "Creates new output files after each init"); |
| WEBRTC_DEFINE_string(custom_call_order_file, |
| "", |
| "Custom process API call order file"); |
| WEBRTC_DEFINE_bool(print_aec3_parameter_values, |
| false, |
| "Print parameter values used in AEC3 in JSON-format"); |
| WEBRTC_DEFINE_string(aec3_settings, |
| "", |
| "File in JSON-format with custom AEC3 settings"); |
| WEBRTC_DEFINE_bool(dump_data, |
| false, |
| "Dump internal data during the call (requires build flag)"); |
| WEBRTC_DEFINE_bool(help, false, "Print this message"); |
| |
| void SetSettingIfSpecified(const std::string& value, |
| absl::optional<std::string>* parameter) { |
| if (value.compare("") != 0) { |
| *parameter = value; |
| } |
| } |
| |
| void SetSettingIfSpecified(int value, absl::optional<int>* parameter) { |
| if (value != kParameterNotSpecifiedValue) { |
| *parameter = value; |
| } |
| } |
| |
| void SetSettingIfFlagSet(int32_t flag, absl::optional<bool>* parameter) { |
| if (flag == 0) { |
| *parameter = false; |
| } else if (flag == 1) { |
| *parameter = true; |
| } |
| } |
| |
| SimulationSettings CreateSettings() { |
| SimulationSettings settings; |
| if (FLAG_all_default) { |
| settings.use_le = true; |
| settings.use_vad = true; |
| settings.use_ie = false; |
| settings.use_ts = true; |
| settings.use_ns = true; |
| settings.use_hpf = true; |
| settings.use_agc = true; |
| settings.use_agc2 = false; |
| settings.use_pre_amplifier = false; |
| settings.use_aec = true; |
| settings.use_aecm = false; |
| settings.use_ed = false; |
| } |
| SetSettingIfSpecified(FLAG_dump_input, &settings.aec_dump_input_filename); |
| SetSettingIfSpecified(FLAG_dump_output, &settings.aec_dump_output_filename); |
| SetSettingIfSpecified(FLAG_i, &settings.input_filename); |
| SetSettingIfSpecified(FLAG_o, &settings.output_filename); |
| SetSettingIfSpecified(FLAG_ri, &settings.reverse_input_filename); |
| SetSettingIfSpecified(FLAG_ro, &settings.reverse_output_filename); |
| SetSettingIfSpecified(FLAG_artificial_nearend, |
| &settings.artificial_nearend_filename); |
| SetSettingIfSpecified(FLAG_output_num_channels, |
| &settings.output_num_channels); |
| SetSettingIfSpecified(FLAG_reverse_output_num_channels, |
| &settings.reverse_output_num_channels); |
| SetSettingIfSpecified(FLAG_output_sample_rate_hz, |
| &settings.output_sample_rate_hz); |
| SetSettingIfSpecified(FLAG_reverse_output_sample_rate_hz, |
| &settings.reverse_output_sample_rate_hz); |
| SetSettingIfFlagSet(FLAG_aec, &settings.use_aec); |
| SetSettingIfFlagSet(FLAG_aecm, &settings.use_aecm); |
| SetSettingIfFlagSet(FLAG_ed, &settings.use_ed); |
| SetSettingIfSpecified(FLAG_ed_graph, &settings.ed_graph_output_filename); |
| SetSettingIfFlagSet(FLAG_agc, &settings.use_agc); |
| SetSettingIfFlagSet(FLAG_agc2, &settings.use_agc2); |
| SetSettingIfFlagSet(FLAG_pre_amplifier, &settings.use_pre_amplifier); |
| SetSettingIfFlagSet(FLAG_hpf, &settings.use_hpf); |
| SetSettingIfFlagSet(FLAG_ns, &settings.use_ns); |
| SetSettingIfFlagSet(FLAG_ts, &settings.use_ts); |
| SetSettingIfFlagSet(FLAG_vad, &settings.use_vad); |
| SetSettingIfFlagSet(FLAG_le, &settings.use_le); |
| SetSettingIfSpecified(FLAG_aec_suppression_level, |
| &settings.aec_suppression_level); |
| SetSettingIfFlagSet(FLAG_delay_agnostic, &settings.use_delay_agnostic); |
| SetSettingIfFlagSet(FLAG_extended_filter, &settings.use_extended_filter); |
| SetSettingIfFlagSet(FLAG_refined_adaptive_filter, |
| &settings.use_refined_adaptive_filter); |
| |
| SetSettingIfFlagSet(FLAG_aec3, &settings.use_aec3); |
| SetSettingIfFlagSet(FLAG_experimental_agc, &settings.use_experimental_agc); |
| SetSettingIfFlagSet(FLAG_experimental_agc_disable_digital_adaptive, |
| &settings.experimental_agc_disable_digital_adaptive); |
| SetSettingIfFlagSet(FLAG_experimental_agc_analyze_before_aec, |
| &settings.experimental_agc_analyze_before_aec); |
| SetSettingIfFlagSet(FLAG_experimental_agc_agc2_level_estimator, |
| &settings.use_experimental_agc_agc2_level_estimator); |
| SetSettingIfSpecified(FLAG_agc_mode, &settings.agc_mode); |
| SetSettingIfSpecified(FLAG_agc_target_level, &settings.agc_target_level); |
| SetSettingIfFlagSet(FLAG_agc_limiter, &settings.use_agc_limiter); |
| SetSettingIfSpecified(FLAG_agc_compression_gain, |
| &settings.agc_compression_gain); |
| SetSettingIfFlagSet(FLAG_agc2_enable_adaptive_gain, |
| &settings.agc2_use_adaptive_gain); |
| settings.agc2_fixed_gain_db = FLAG_agc2_fixed_gain_db; |
| settings.pre_amplifier_gain_factor = FLAG_pre_amplifier_gain_factor; |
| SetSettingIfSpecified(FLAG_vad_likelihood, &settings.vad_likelihood); |
| SetSettingIfSpecified(FLAG_ns_level, &settings.ns_level); |
| SetSettingIfSpecified(FLAG_stream_delay, &settings.stream_delay); |
| SetSettingIfFlagSet(FLAG_use_stream_delay, &settings.use_stream_delay); |
| SetSettingIfSpecified(FLAG_stream_drift_samples, |
| &settings.stream_drift_samples); |
| SetSettingIfSpecified(FLAG_custom_call_order_file, |
| &settings.custom_call_order_filename); |
| SetSettingIfSpecified(FLAG_aec3_settings, &settings.aec3_settings_filename); |
| settings.initial_mic_level = FLAG_initial_mic_level; |
| settings.simulate_mic_gain = FLAG_simulate_mic_gain; |
| SetSettingIfSpecified(FLAG_simulated_mic_kind, &settings.simulated_mic_kind); |
| settings.report_performance = FLAG_performance_report; |
| settings.use_verbose_logging = FLAG_verbose; |
| settings.use_quiet_output = FLAG_quiet; |
| settings.report_bitexactness = FLAG_bitexactness_report; |
| settings.discard_all_settings_in_aecdump = FLAG_discard_settings_in_aecdump; |
| settings.fixed_interface = FLAG_fixed_interface; |
| settings.store_intermediate_output = FLAG_store_intermediate_output; |
| settings.print_aec3_parameter_values = FLAG_print_aec3_parameter_values; |
| settings.dump_internal_data = FLAG_dump_data; |
| |
| return settings; |
| } |
| |
| void ReportConditionalErrorAndExit(bool condition, const std::string& message) { |
| if (condition) { |
| std::cerr << message << std::endl; |
| exit(1); |
| } |
| } |
| |
| void PerformBasicParameterSanityChecks(const SimulationSettings& settings) { |
| if (settings.input_filename || settings.reverse_input_filename) { |
| ReportConditionalErrorAndExit(!!settings.aec_dump_input_filename, |
| "Error: The aec dump cannot be specified " |
| "together with input wav files!\n"); |
| |
| ReportConditionalErrorAndExit(!!settings.artificial_nearend_filename, |
| "Error: The artificial nearend cannot be " |
| "specified together with input wav files!\n"); |
| |
| ReportConditionalErrorAndExit(!settings.input_filename, |
| "Error: When operating at wav files, the " |
| "input wav filename must be " |
| "specified!\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.reverse_output_filename && !settings.reverse_input_filename, |
| "Error: When operating at wav files, the reverse input wav filename " |
| "must be specified if the reverse output wav filename is specified!\n"); |
| } else { |
| ReportConditionalErrorAndExit(!settings.aec_dump_input_filename, |
| "Error: Either the aec dump or the wav " |
| "input files must be specified!\n"); |
| } |
| |
| ReportConditionalErrorAndExit( |
| settings.use_aec && *settings.use_aec && settings.use_aecm && |
| *settings.use_aecm, |
| "Error: The AEC and the AECM cannot be activated at the same time!\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.output_sample_rate_hz && *settings.output_sample_rate_hz <= 0, |
| "Error: --output_sample_rate_hz must be positive!\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.reverse_output_sample_rate_hz && |
| settings.output_sample_rate_hz && |
| *settings.output_sample_rate_hz <= 0, |
| "Error: --reverse_output_sample_rate_hz must be positive!\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.output_num_channels && *settings.output_num_channels <= 0, |
| "Error: --output_num_channels must be positive!\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.reverse_output_num_channels && |
| *settings.reverse_output_num_channels <= 0, |
| "Error: --reverse_output_num_channels must be positive!\n"); |
| |
| ReportConditionalErrorAndExit(settings.aec_suppression_level && |
| ((*settings.aec_suppression_level) < 1 || |
| (*settings.aec_suppression_level) > 2), |
| "Error: --aec_suppression_level must be " |
| "specified between 1 and 2. 0 is " |
| "deprecated.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.agc_target_level && ((*settings.agc_target_level) < 0 || |
| (*settings.agc_target_level) > 31), |
| "Error: --agc_target_level must be specified between 0 and 31.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.agc_compression_gain && ((*settings.agc_compression_gain) < 0 || |
| (*settings.agc_compression_gain) > 90), |
| "Error: --agc_compression_gain must be specified between 0 and 90.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.use_agc2 && *settings.use_agc2 && |
| ((settings.agc2_fixed_gain_db) < 0 || |
| (settings.agc2_fixed_gain_db) > 90), |
| "Error: --agc2_fixed_gain_db must be specified between 0 and 90.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.vad_likelihood && |
| ((*settings.vad_likelihood) < 0 || (*settings.vad_likelihood) > 3), |
| "Error: --vad_likelihood must be specified between 0 and 3.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.ns_level && |
| ((*settings.ns_level) < 0 || (*settings.ns_level) > 3), |
| "Error: --ns_level must be specified between 0 and 3.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.report_bitexactness && !settings.aec_dump_input_filename, |
| "Error: --bitexactness_report can only be used when operating on an " |
| "aecdump\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.custom_call_order_filename && settings.aec_dump_input_filename, |
| "Error: --custom_call_order_file cannot be used when operating on an " |
| "aecdump\n"); |
| |
| ReportConditionalErrorAndExit( |
| (settings.initial_mic_level < 0 || settings.initial_mic_level > 255), |
| "Error: --initial_mic_level must be specified between 0 and 255.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.simulated_mic_kind && !settings.simulate_mic_gain, |
| "Error: --simulated_mic_kind cannot be specified when mic simulation is " |
| "disabled\n"); |
| |
| ReportConditionalErrorAndExit( |
| !settings.simulated_mic_kind && settings.simulate_mic_gain, |
| "Error: --simulated_mic_kind must be specified when mic simulation is " |
| "enabled\n"); |
| |
| auto valid_wav_name = [](const std::string& wav_file_name) { |
| if (wav_file_name.size() < 5) { |
| return false; |
| } |
| if ((wav_file_name.compare(wav_file_name.size() - 4, 4, ".wav") == 0) || |
| (wav_file_name.compare(wav_file_name.size() - 4, 4, ".WAV") == 0)) { |
| return true; |
| } |
| return false; |
| }; |
| |
| ReportConditionalErrorAndExit( |
| settings.input_filename && (!valid_wav_name(*settings.input_filename)), |
| "Error: --i must be a valid .wav file name.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.output_filename && (!valid_wav_name(*settings.output_filename)), |
| "Error: --o must be a valid .wav file name.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.reverse_input_filename && |
| (!valid_wav_name(*settings.reverse_input_filename)), |
| "Error: --ri must be a valid .wav file name.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.reverse_output_filename && |
| (!valid_wav_name(*settings.reverse_output_filename)), |
| "Error: --ro must be a valid .wav file name.\n"); |
| |
| ReportConditionalErrorAndExit( |
| settings.artificial_nearend_filename && |
| !valid_wav_name(*settings.artificial_nearend_filename), |
| "Error: --artifical_nearend must be a valid .wav file name.\n"); |
| |
| ReportConditionalErrorAndExit( |
| WEBRTC_APM_DEBUG_DUMP == 0 && settings.dump_internal_data, |
| "Error: --dump_data cannot be set without proper build support.\n"); |
| } |
| |
| } // namespace |
| |
| int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder, |
| int argc, |
| char* argv[]) { |
| if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help || |
| argc != 1) { |
| printf("%s", kUsageDescription); |
| if (FLAG_help) { |
| rtc::FlagList::Print(nullptr, false); |
| return 0; |
| } |
| return 1; |
| } |
| |
| SimulationSettings settings = CreateSettings(); |
| PerformBasicParameterSanityChecks(settings); |
| std::unique_ptr<AudioProcessingSimulator> processor; |
| |
| if (settings.aec_dump_input_filename) { |
| processor.reset(new AecDumpBasedSimulator(settings, std::move(ap_builder))); |
| } else { |
| processor.reset(new WavBasedSimulator(settings, std::move(ap_builder))); |
| } |
| |
| processor->Process(); |
| |
| if (settings.report_performance) { |
| const auto& proc_time = processor->proc_time(); |
| int64_t exec_time_us = proc_time.sum / rtc::kNumNanosecsPerMicrosec; |
| std::cout << std::endl |
| << "Execution time: " << exec_time_us * 1e-6 << " s, File time: " |
| << processor->get_num_process_stream_calls() * 1.f / |
| AudioProcessingSimulator::kChunksPerSecond |
| << std::endl |
| << "Time per fwd stream chunk (mean, max, min): " << std::endl |
| << exec_time_us * 1.f / processor->get_num_process_stream_calls() |
| << " us, " << 1.f * proc_time.max / rtc::kNumNanosecsPerMicrosec |
| << " us, " << 1.f * proc_time.min / rtc::kNumNanosecsPerMicrosec |
| << " us" << std::endl; |
| } |
| |
| if (settings.report_bitexactness && settings.aec_dump_input_filename) { |
| if (processor->OutputWasBitexact()) { |
| std::cout << "The processing was bitexact."; |
| } else { |
| std::cout << "The processing was not bitexact."; |
| } |
| } |
| |
| return 0; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |
