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webrtc/modules/audio_processing/test/audioproc_float_impl.cc
Mirko Bonadei 5ccdc1331f Prefix flag macros with WEBRTC_. 
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +00:00

520 lines
22 KiB
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/*
* 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(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;
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");
}
} // 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
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