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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc | 821 |
1 files changed, 821 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc b/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc new file mode 100644 index 0000000000..c23ec74366 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc @@ -0,0 +1,821 @@ +/* + * 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 "modules/audio_processing/test/audioproc_float_impl.h" + +#include <string.h> + +#include <iostream> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "absl/strings/string_view.h" +#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/wav_based_simulator.h" +#include "rtc_base/checks.h" +#include "rtc_base/strings/string_builder.h" +#include "system_wrappers/include/field_trial.h" + +constexpr int kParameterNotSpecifiedValue = -10000; + +ABSL_FLAG(std::string, dump_input, "", "Aec dump input filename"); +ABSL_FLAG(std::string, dump_output, "", "Aec dump output filename"); +ABSL_FLAG(std::string, i, "", "Forward stream input wav filename"); +ABSL_FLAG(std::string, o, "", "Forward stream output wav filename"); +ABSL_FLAG(std::string, ri, "", "Reverse stream input wav filename"); +ABSL_FLAG(std::string, ro, "", "Reverse stream output wav filename"); +ABSL_FLAG(std::string, + artificial_nearend, + "", + "Artificial nearend wav filename"); +ABSL_FLAG(std::string, linear_aec_output, "", "Linear AEC output wav filename"); +ABSL_FLAG(int, + output_num_channels, + kParameterNotSpecifiedValue, + "Number of forward stream output channels"); +ABSL_FLAG(int, + reverse_output_num_channels, + kParameterNotSpecifiedValue, + "Number of Reverse stream output channels"); +ABSL_FLAG(int, + output_sample_rate_hz, + kParameterNotSpecifiedValue, + "Forward stream output sample rate in Hz"); +ABSL_FLAG(int, + reverse_output_sample_rate_hz, + kParameterNotSpecifiedValue, + "Reverse stream output sample rate in Hz"); +ABSL_FLAG(bool, + fixed_interface, + false, + "Use the fixed interface when operating on wav files"); +ABSL_FLAG(int, + aec, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the echo canceller"); +ABSL_FLAG(int, + aecm, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the mobile echo controller"); +ABSL_FLAG(int, + ed, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the residual echo detector"); +ABSL_FLAG(std::string, + ed_graph, + "", + "Output filename for graph of echo likelihood"); +ABSL_FLAG(int, + agc, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the AGC"); +ABSL_FLAG(int, + agc2, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the AGC2"); +ABSL_FLAG(int, + pre_amplifier, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the pre amplifier"); +ABSL_FLAG( + int, + capture_level_adjustment, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the capture level adjustment functionality"); +ABSL_FLAG(int, + analog_mic_gain_emulation, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate(0) the analog mic gain emulation in the " + "production (non-test) code."); +ABSL_FLAG(int, + hpf, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the high-pass filter"); +ABSL_FLAG(int, + ns, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the noise suppressor"); +ABSL_FLAG(int, + ts, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the transient suppressor"); +ABSL_FLAG(int, + analog_agc, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the analog AGC"); +ABSL_FLAG(bool, + all_default, + false, + "Activate all of the default components (will be overridden by any " + "other settings)"); +ABSL_FLAG(int, + analog_agc_use_digital_adaptive_controller, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) digital adaptation in AGC1. " + "Digital adaptation is active by default."); +ABSL_FLAG(int, + agc_mode, + kParameterNotSpecifiedValue, + "Specify the AGC mode (0-2)"); +ABSL_FLAG(int, + agc_target_level, + kParameterNotSpecifiedValue, + "Specify the AGC target level (0-31)"); +ABSL_FLAG(int, + agc_limiter, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the level estimator"); +ABSL_FLAG(int, + agc_compression_gain, + kParameterNotSpecifiedValue, + "Specify the AGC compression gain (0-90)"); +ABSL_FLAG(int, + agc2_enable_adaptive_gain, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) the AGC2 adaptive gain"); +ABSL_FLAG(float, + agc2_fixed_gain_db, + kParameterNotSpecifiedValue, + "AGC2 fixed gain (dB) to apply"); +ABSL_FLAG(float, + pre_amplifier_gain_factor, + kParameterNotSpecifiedValue, + "Pre-amplifier gain factor (linear) to apply"); +ABSL_FLAG(float, + pre_gain_factor, + kParameterNotSpecifiedValue, + "Pre-gain factor (linear) to apply in the capture level adjustment"); +ABSL_FLAG(float, + post_gain_factor, + kParameterNotSpecifiedValue, + "Post-gain factor (linear) to apply in the capture level adjustment"); +ABSL_FLAG(float, + analog_mic_gain_emulation_initial_level, + kParameterNotSpecifiedValue, + "Emulated analog mic level to apply initially in the production " + "(non-test) code."); +ABSL_FLAG(int, + ns_level, + kParameterNotSpecifiedValue, + "Specify the NS level (0-3)"); +ABSL_FLAG(int, + ns_analysis_on_linear_aec_output, + kParameterNotSpecifiedValue, + "Specifies whether the noise suppression analysis is done on the " + "linear AEC output"); +ABSL_FLAG(int, + maximum_internal_processing_rate, + kParameterNotSpecifiedValue, + "Set a maximum internal processing rate (32000 or 48000) to override " + "the default rate"); +ABSL_FLAG(int, + stream_delay, + kParameterNotSpecifiedValue, + "Specify the stream delay in ms to use"); +ABSL_FLAG(int, + use_stream_delay, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) reporting the stream delay"); +ABSL_FLAG(int, + stream_drift_samples, + kParameterNotSpecifiedValue, + "Specify the number of stream drift samples to use"); +ABSL_FLAG(int, + initial_mic_level, + 100, + "Initial mic level (0-255) for the analog mic gain simulation in the " + "test code"); +ABSL_FLAG(int, + simulate_mic_gain, + 0, + "Activate (1) or deactivate(0) the analog mic gain simulation in the " + "test code"); +ABSL_FLAG(int, + multi_channel_render, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) multi-channel render processing in " + "APM pipeline"); +ABSL_FLAG(int, + multi_channel_capture, + kParameterNotSpecifiedValue, + "Activate (1) or deactivate (0) multi-channel capture processing in " + "APM pipeline"); +ABSL_FLAG(int, + simulated_mic_kind, + kParameterNotSpecifiedValue, + "Specify which microphone kind to use for microphone simulation"); +ABSL_FLAG(int, + override_key_pressed, + kParameterNotSpecifiedValue, + "Always set to true (1) or to false (0) the key press state. If " + "unspecified, false is set with Wav files or, with AEC dumps, the " + "recorded event is used."); +ABSL_FLAG(int, + frame_for_sending_capture_output_used_false, + kParameterNotSpecifiedValue, + "Capture frame index for sending a runtime setting for that the " + "capture output is not used."); +ABSL_FLAG(int, + frame_for_sending_capture_output_used_true, + kParameterNotSpecifiedValue, + "Capture frame index for sending a runtime setting for that the " + "capture output is used."); +ABSL_FLAG(bool, performance_report, false, "Report the APM performance "); +ABSL_FLAG(std::string, + performance_report_output_file, + "", + "Generate a CSV file with the API call durations"); +ABSL_FLAG(bool, verbose, false, "Produce verbose output"); +ABSL_FLAG(bool, + quiet, + false, + "Avoid producing information about the progress."); +ABSL_FLAG(bool, + bitexactness_report, + false, + "Report bitexactness for aec dump result reproduction"); +ABSL_FLAG(bool, + discard_settings_in_aecdump, + false, + "Discard any config settings specified in the aec dump"); +ABSL_FLAG(bool, + store_intermediate_output, + false, + "Creates new output files after each init"); +ABSL_FLAG(std::string, + custom_call_order_file, + "", + "Custom process API call order file"); +ABSL_FLAG(std::string, + output_custom_call_order_file, + "", + "Generate custom process API call order file from AEC dump"); +ABSL_FLAG(bool, + print_aec_parameter_values, + false, + "Print parameter values used in AEC in JSON-format"); +ABSL_FLAG(std::string, + aec_settings, + "", + "File in JSON-format with custom AEC settings"); +ABSL_FLAG(bool, + dump_data, + false, + "Dump internal data during the call (requires build flag)"); +ABSL_FLAG(std::string, + dump_data_output_dir, + "", + "Internal data dump output directory"); +ABSL_FLAG(int, + dump_set_to_use, + kParameterNotSpecifiedValue, + "Specifies the dump set to use (if not all the dump sets will " + "be used"); +ABSL_FLAG(bool, + analyze, + false, + "Only analyze the call setup behavior (no processing)"); +ABSL_FLAG(float, + dump_start_seconds, + kParameterNotSpecifiedValue, + "Start of when to dump data (seconds)."); +ABSL_FLAG(float, + dump_end_seconds, + kParameterNotSpecifiedValue, + "End of when to dump data (seconds)."); +ABSL_FLAG(int, + dump_start_frame, + kParameterNotSpecifiedValue, + "Start of when to dump data (frames)."); +ABSL_FLAG(int, + dump_end_frame, + kParameterNotSpecifiedValue, + "End of when to dump data (frames)."); +ABSL_FLAG(int, + init_to_process, + kParameterNotSpecifiedValue, + "Init index to process."); + +ABSL_FLAG(bool, + float_wav_output, + false, + "Produce floating point wav output files."); + +ABSL_FLAG(std::string, + force_fieldtrials, + "", + "Field trials control experimental feature code which can be forced. " + "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/" + " will assign the group Enable to field trial WebRTC-FooFeature."); + +namespace webrtc { +namespace test { +namespace { + +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"; + +void SetSettingIfSpecified(absl::string_view value, + absl::optional<std::string>* parameter) { + if (value.compare("") != 0) { + *parameter = std::string(value); + } +} + +void SetSettingIfSpecified(int value, absl::optional<int>* parameter) { + if (value != kParameterNotSpecifiedValue) { + *parameter = value; + } +} + +void SetSettingIfSpecified(float value, absl::optional<float>* parameter) { + constexpr float kFloatParameterNotSpecifiedValue = + kParameterNotSpecifiedValue; + if (value != kFloatParameterNotSpecifiedValue) { + *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 (absl::GetFlag(FLAGS_all_default)) { + settings.use_ts = true; + settings.use_analog_agc = 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(absl::GetFlag(FLAGS_dump_input), + &settings.aec_dump_input_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_output), + &settings.aec_dump_output_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_i), &settings.input_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_o), &settings.output_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_ri), + &settings.reverse_input_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_ro), + &settings.reverse_output_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_artificial_nearend), + &settings.artificial_nearend_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_linear_aec_output), + &settings.linear_aec_output_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_output_num_channels), + &settings.output_num_channels); + SetSettingIfSpecified(absl::GetFlag(FLAGS_reverse_output_num_channels), + &settings.reverse_output_num_channels); + SetSettingIfSpecified(absl::GetFlag(FLAGS_output_sample_rate_hz), + &settings.output_sample_rate_hz); + SetSettingIfSpecified(absl::GetFlag(FLAGS_reverse_output_sample_rate_hz), + &settings.reverse_output_sample_rate_hz); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_aec), &settings.use_aec); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_aecm), &settings.use_aecm); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_ed), &settings.use_ed); + SetSettingIfSpecified(absl::GetFlag(FLAGS_ed_graph), + &settings.ed_graph_output_filename); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc), &settings.use_agc); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc2), &settings.use_agc2); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_pre_amplifier), + &settings.use_pre_amplifier); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_capture_level_adjustment), + &settings.use_capture_level_adjustment); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_mic_gain_emulation), + &settings.use_analog_mic_gain_emulation); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_hpf), &settings.use_hpf); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_ns), &settings.use_ns); + SetSettingIfSpecified(absl::GetFlag(FLAGS_ts), &settings.use_ts); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc), + &settings.use_analog_agc); + SetSettingIfFlagSet( + absl::GetFlag(FLAGS_analog_agc_use_digital_adaptive_controller), + &settings.analog_agc_use_digital_adaptive_controller); + SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_mode), &settings.agc_mode); + SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_target_level), + &settings.agc_target_level); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc_limiter), + &settings.use_agc_limiter); + SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_compression_gain), + &settings.agc_compression_gain); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc2_enable_adaptive_gain), + &settings.agc2_use_adaptive_gain); + + SetSettingIfSpecified(absl::GetFlag(FLAGS_agc2_fixed_gain_db), + &settings.agc2_fixed_gain_db); + SetSettingIfSpecified(absl::GetFlag(FLAGS_pre_amplifier_gain_factor), + &settings.pre_amplifier_gain_factor); + SetSettingIfSpecified(absl::GetFlag(FLAGS_pre_gain_factor), + &settings.pre_gain_factor); + SetSettingIfSpecified(absl::GetFlag(FLAGS_post_gain_factor), + &settings.post_gain_factor); + SetSettingIfSpecified( + absl::GetFlag(FLAGS_analog_mic_gain_emulation_initial_level), + &settings.analog_mic_gain_emulation_initial_level); + SetSettingIfSpecified(absl::GetFlag(FLAGS_ns_level), &settings.ns_level); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_ns_analysis_on_linear_aec_output), + &settings.ns_analysis_on_linear_aec_output); + SetSettingIfSpecified(absl::GetFlag(FLAGS_maximum_internal_processing_rate), + &settings.maximum_internal_processing_rate); + SetSettingIfSpecified(absl::GetFlag(FLAGS_stream_delay), + &settings.stream_delay); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_use_stream_delay), + &settings.use_stream_delay); + SetSettingIfSpecified(absl::GetFlag(FLAGS_custom_call_order_file), + &settings.call_order_input_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_output_custom_call_order_file), + &settings.call_order_output_filename); + SetSettingIfSpecified(absl::GetFlag(FLAGS_aec_settings), + &settings.aec_settings_filename); + settings.initial_mic_level = absl::GetFlag(FLAGS_initial_mic_level); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_render), + &settings.multi_channel_render); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_capture), + &settings.multi_channel_capture); + settings.simulate_mic_gain = absl::GetFlag(FLAGS_simulate_mic_gain); + SetSettingIfSpecified(absl::GetFlag(FLAGS_simulated_mic_kind), + &settings.simulated_mic_kind); + SetSettingIfFlagSet(absl::GetFlag(FLAGS_override_key_pressed), + &settings.override_key_pressed); + SetSettingIfSpecified( + absl::GetFlag(FLAGS_frame_for_sending_capture_output_used_false), + &settings.frame_for_sending_capture_output_used_false); + SetSettingIfSpecified( + absl::GetFlag(FLAGS_frame_for_sending_capture_output_used_true), + &settings.frame_for_sending_capture_output_used_true); + settings.report_performance = absl::GetFlag(FLAGS_performance_report); + SetSettingIfSpecified(absl::GetFlag(FLAGS_performance_report_output_file), + &settings.performance_report_output_filename); + settings.use_verbose_logging = absl::GetFlag(FLAGS_verbose); + settings.use_quiet_output = absl::GetFlag(FLAGS_quiet); + settings.report_bitexactness = absl::GetFlag(FLAGS_bitexactness_report); + settings.discard_all_settings_in_aecdump = + absl::GetFlag(FLAGS_discard_settings_in_aecdump); + settings.fixed_interface = absl::GetFlag(FLAGS_fixed_interface); + settings.store_intermediate_output = + absl::GetFlag(FLAGS_store_intermediate_output); + settings.print_aec_parameter_values = + absl::GetFlag(FLAGS_print_aec_parameter_values); + settings.dump_internal_data = absl::GetFlag(FLAGS_dump_data); + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_data_output_dir), + &settings.dump_internal_data_output_dir); + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_set_to_use), + &settings.dump_set_to_use); + settings.wav_output_format = absl::GetFlag(FLAGS_float_wav_output) + ? WavFile::SampleFormat::kFloat + : WavFile::SampleFormat::kInt16; + + settings.analysis_only = absl::GetFlag(FLAGS_analyze); + + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_start_frame), + &settings.dump_start_frame); + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_end_frame), + &settings.dump_end_frame); + + constexpr int kFramesPerSecond = 100; + absl::optional<float> start_seconds; + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_start_seconds), + &start_seconds); + if (start_seconds) { + settings.dump_start_frame = *start_seconds * kFramesPerSecond; + } + + absl::optional<float> end_seconds; + SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_end_seconds), &end_seconds); + if (end_seconds) { + settings.dump_end_frame = *end_seconds * kFramesPerSecond; + } + + SetSettingIfSpecified(absl::GetFlag(FLAGS_init_to_process), + &settings.init_to_process); + + return settings; +} + +void ReportConditionalErrorAndExit(bool condition, absl::string_view message) { + if (condition) { + std::cerr << message << std::endl; + exit(1); + } +} + +void PerformBasicParameterSanityChecks( + const SimulationSettings& settings, + bool pre_constructed_ap_provided, + bool pre_constructed_ap_builder_provided) { + if (settings.input_filename || settings.reverse_input_filename) { + ReportConditionalErrorAndExit( + !!settings.aec_dump_input_filename, + "Error: The aec dump file cannot be specified " + "together with input wav files!\n"); + + ReportConditionalErrorAndExit( + !!settings.aec_dump_input_string, + "Error: The aec dump input string 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 && !settings.aec_dump_input_string, + "Error: Either the aec dump input file, the wav " + "input file or the aec dump input string must be specified!\n"); + ReportConditionalErrorAndExit( + settings.aec_dump_input_filename && settings.aec_dump_input_string, + "Error: The aec dump input file cannot be specified together with the " + "aec dump input string!\n"); + } + + ReportConditionalErrorAndExit(settings.use_aec && !(*settings.use_aec) && + settings.linear_aec_output_filename, + "Error: The linear AEC ouput filename cannot " + "be specified without the AEC being active"); + + 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.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.agc2_fixed_gain_db && ((*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.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.call_order_input_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"); + + // TODO(bugs.webrtc.org/7494): Document how the two settings below differ. + ReportConditionalErrorAndExit( + settings.simulate_mic_gain && settings.use_analog_mic_gain_emulation, + "Error: --simulate_mic_gain and --use_analog_mic_gain_emulation cannot " + "be enabled at the same time\n"); + + auto valid_wav_name = [](absl::string_view 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( + settings.linear_aec_output_filename && + (!valid_wav_name(*settings.linear_aec_output_filename)), + "Error: --linear_aec_output 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"); + + ReportConditionalErrorAndExit(settings.init_to_process && + *settings.init_to_process != 1 && + !settings.aec_dump_input_filename, + "Error: --init_to_process must be set to 1 for " + "wav-file based simulations.\n"); + + ReportConditionalErrorAndExit( + !settings.init_to_process && + (settings.dump_start_frame || settings.dump_end_frame), + "Error: --init_to_process must be set when specifying a start and/or end " + "frame for when to dump internal data.\n"); + + ReportConditionalErrorAndExit( + !settings.dump_internal_data && + settings.dump_internal_data_output_dir.has_value(), + "Error: --dump_data_output_dir cannot be set without --dump_data.\n"); + + ReportConditionalErrorAndExit( + !settings.aec_dump_input_filename && + settings.call_order_output_filename.has_value(), + "Error: --output_custom_call_order_file needs an AEC dump input file.\n"); + + ReportConditionalErrorAndExit( + (!settings.use_pre_amplifier || !(*settings.use_pre_amplifier)) && + settings.pre_amplifier_gain_factor.has_value(), + "Error: --pre_amplifier_gain_factor needs --pre_amplifier to be " + "specified and set.\n"); + + ReportConditionalErrorAndExit( + pre_constructed_ap_provided && pre_constructed_ap_builder_provided, + "Error: The AudioProcessing and the AudioProcessingBuilder cannot both " + "be specified at the same time.\n"); + + ReportConditionalErrorAndExit( + settings.aec_settings_filename && pre_constructed_ap_provided, + "Error: The aec_settings_filename cannot be specified when a " + "pre-constructed audio processing object is provided.\n"); + + ReportConditionalErrorAndExit( + settings.aec_settings_filename && pre_constructed_ap_provided, + "Error: The print_aec_parameter_values cannot be set when a " + "pre-constructed audio processing object is provided.\n"); + + if (settings.linear_aec_output_filename && pre_constructed_ap_provided) { + std::cout << "Warning: For the linear AEC output to be stored, this must " + "be configured in the AEC that is part of the provided " + "AudioProcessing object." + << std::endl; + } +} + +int RunSimulation(rtc::scoped_refptr<AudioProcessing> audio_processing, + std::unique_ptr<AudioProcessingBuilder> ap_builder, + int argc, + char* argv[], + absl::string_view input_aecdump, + std::vector<float>* processed_capture_samples) { + std::vector<char*> args = absl::ParseCommandLine(argc, argv); + if (args.size() != 1) { + printf("%s", kUsageDescription); + return 1; + } + // InitFieldTrialsFromString stores the char*, so the char array must + // outlive the application. + const std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials); + webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str()); + + SimulationSettings settings = CreateSettings(); + if (!input_aecdump.empty()) { + settings.aec_dump_input_string = input_aecdump; + settings.processed_capture_samples = processed_capture_samples; + RTC_CHECK(settings.processed_capture_samples); + } + PerformBasicParameterSanityChecks(settings, !!audio_processing, !!ap_builder); + std::unique_ptr<AudioProcessingSimulator> processor; + + if (settings.aec_dump_input_filename || settings.aec_dump_input_string) { + processor.reset(new AecDumpBasedSimulator( + settings, std::move(audio_processing), std::move(ap_builder))); + } else { + processor.reset(new WavBasedSimulator(settings, std::move(audio_processing), + std::move(ap_builder))); + } + + if (settings.analysis_only) { + processor->Analyze(); + } else { + processor->Process(); + } + + if (settings.report_performance) { + processor->GetApiCallStatistics().PrintReport(); + } + if (settings.performance_report_output_filename) { + processor->GetApiCallStatistics().WriteReportToFile( + *settings.performance_report_output_filename); + } + + 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 + +int AudioprocFloatImpl(rtc::scoped_refptr<AudioProcessing> audio_processing, + int argc, + char* argv[]) { + return RunSimulation( + std::move(audio_processing), /*ap_builder=*/nullptr, argc, argv, + /*input_aecdump=*/"", /*processed_capture_samples=*/nullptr); +} + +int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder, + int argc, + char* argv[], + absl::string_view input_aecdump, + std::vector<float>* processed_capture_samples) { + return RunSimulation(/*audio_processing=*/nullptr, std::move(ap_builder), + argc, argv, input_aecdump, processed_capture_samples); +} + +} // namespace test +} // namespace webrtc |