mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_processing/test/audioproc_float_impl.cc
Per Åhgren db5d728878 Add refined handling of the internal scaling of the audio in APM 
This CL adds functionality that allows adjusting the audio levels
internally in APM. The main purpose of the functionality is to allow
APM to optionally be moved to an integration that does not provide an
analog gain to control, and the implementation of this has been
tailored specifically to meet the requirements for that.

More specifically, this CL does
-Add a new variant of the pre-amplifier gain that is intended to replace
 the pre-amplifier gain (but at the moment can coexist with that). The
 main differences with the pre-amplifier gain is that an attenuating
 gain is allowed, the gain is applied jointly with any emulated analog
 gain, and that its packaging fits better with the post gain.
-Add an emulation of an analog microphone gain. The emulation is
 designed to match the analog mic gain functionality in Chrome OS (which
 is digital) but should be usable also on other platforms.
-Add a post-gain which is applied after all processing has been applied.
 The purpose of this gain is for it to work well with the integration
 in ChromeOS, and be used to compensate for the offset that there is
 applied on some USB audio devices.


Bug: b/177830918
Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33466}
2021-03-15 19:12:02 +00:00

850 lines
34 KiB
C++
Raw Blame History

/*
* 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), deactivate (0) or activate the transient suppressor "
"with continuous key events (2)");
ABSL_FLAG(int,
analog_agc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate (0) the analog AGC");
ABSL_FLAG(int,
vad,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate (0) the voice activity detector");
ABSL_FLAG(int,
le,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate (0) the level estimator");
ABSL_FLAG(bool,
all_default,
false,
"Activate all of the default components (will be overridden by any "
"other settings)");
ABSL_FLAG(int,
analog_agc_disable_digital_adaptive,
kParameterNotSpecifiedValue,
"Force-deactivate (1) digital adaptation in "
"experimental AGC. Digital adaptation is active by default (0).");
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(std::string,
agc2_adaptive_level_estimator,
"RMS",
"AGC2 adaptive digital level estimator to use [RMS, peak]");
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,
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";
std::vector<std::string> GetAgc2AdaptiveLevelEstimatorNames() {
return {"RMS", "peak"};
}
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 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;
}
}
AudioProcessing::Config::GainController2::LevelEstimator
MapAgc2AdaptiveLevelEstimator(absl::string_view name) {
if (name.compare("RMS") == 0) {
return AudioProcessing::Config::GainController2::LevelEstimator::kRms;
}
if (name.compare("peak") == 0) {
return AudioProcessing::Config::GainController2::LevelEstimator::kPeak;
}
auto concat_strings =
[](const std::vector<std::string>& strings) -> std::string {
rtc::StringBuilder ss;
for (const auto& s : strings) {
ss << " " << s;
}
return ss.Release();
};
RTC_CHECK(false)
<< "Invalid value for agc2_adaptive_level_estimator, valid options:"
<< concat_strings(GetAgc2AdaptiveLevelEstimatorNames()) << ".";
}
SimulationSettings CreateSettings() {
SimulationSettings settings;
if (absl::GetFlag(FLAGS_all_default)) {
settings.use_le = true;
settings.use_vad = true;
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_vad), &settings.use_vad);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_le), &settings.use_le);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc_disable_digital_adaptive),
&settings.analog_agc_disable_digital_adaptive);
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);
settings.agc2_adaptive_level_estimator = MapAgc2AdaptiveLevelEstimator(
absl::GetFlag(FLAGS_agc2_adaptive_level_estimator));
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);
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, const std::string& 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");
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(
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
Reference in a new issue View git blame Copy permalink