summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc
diff options
context:
space:
mode:
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.cc821
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