From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../test/py_quality_assessment/BUILD.gn | 170 +++++++ .../test/py_quality_assessment/OWNERS | 5 + .../test/py_quality_assessment/README.md | 125 +++++ .../py_quality_assessment/apm_configs/default.json | 1 + .../apm_quality_assessment.py | 217 +++++++++ .../apm_quality_assessment.sh | 91 ++++ .../apm_quality_assessment_boxplot.py | 154 ++++++ .../apm_quality_assessment_export.py | 63 +++ .../apm_quality_assessment_gencfgs.py | 128 +++++ .../apm_quality_assessment_optimize.py | 189 ++++++++ .../apm_quality_assessment_unittest.py | 28 ++ .../test/py_quality_assessment/output/README.md | 1 + .../quality_assessment/__init__.py | 7 + .../quality_assessment/annotations.py | 296 ++++++++++++ .../quality_assessment/annotations_unittest.py | 160 +++++++ .../quality_assessment/apm_configs/default.json | 1 + .../quality_assessment/apm_vad.cc | 96 ++++ .../quality_assessment/audioproc_wrapper.py | 100 ++++ .../quality_assessment/collect_data.py | 243 ++++++++++ .../quality_assessment/data_access.py | 154 ++++++ .../quality_assessment/echo_path_simulation.py | 136 ++++++ .../echo_path_simulation_factory.py | 48 ++ .../echo_path_simulation_unittest.py | 82 ++++ .../quality_assessment/eval_scores.py | 427 +++++++++++++++++ .../quality_assessment/eval_scores_factory.py | 55 +++ .../quality_assessment/eval_scores_unittest.py | 137 ++++++ .../quality_assessment/evaluation.py | 57 +++ .../quality_assessment/exceptions.py | 45 ++ .../quality_assessment/export.py | 426 +++++++++++++++++ .../quality_assessment/export_unittest.py | 86 ++++ .../quality_assessment/external_vad.py | 75 +++ .../quality_assessment/fake_external_vad.py | 25 + .../quality_assessment/fake_polqa.cc | 56 +++ .../quality_assessment/input_mixer.py | 97 ++++ .../quality_assessment/input_mixer_unittest.py | 140 ++++++ .../quality_assessment/input_signal_creator.py | 68 +++ .../quality_assessment/results.css | 32 ++ .../quality_assessment/results.js | 376 +++++++++++++++ .../quality_assessment/signal_processing.py | 359 ++++++++++++++ .../signal_processing_unittest.py | 183 +++++++ .../quality_assessment/simulation.py | 446 +++++++++++++++++ .../quality_assessment/simulation_unittest.py | 203 ++++++++ .../quality_assessment/sound_level.cc | 127 +++++ .../quality_assessment/test_data_generation.py | 526 +++++++++++++++++++++ .../test_data_generation_factory.py | 71 +++ .../test_data_generation_unittest.py | 207 ++++++++ .../quality_assessment/vad.cc | 103 ++++ 47 files changed, 6822 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json create mode 100755 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py create mode 100755 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py create mode 100755 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py create mode 100755 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py create mode 100755 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py create mode 100644 third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc (limited to 'third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment') diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn new file mode 100644 index 0000000000..e53a829623 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn @@ -0,0 +1,170 @@ +# Copyright (c) 2017 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. + +import("../../../../webrtc.gni") + +if (!build_with_chromium) { + group("py_quality_assessment") { + testonly = true + deps = [ + ":scripts", + ":unit_tests", + ] + } + + copy("scripts") { + testonly = true + sources = [ + "README.md", + "apm_quality_assessment.py", + "apm_quality_assessment.sh", + "apm_quality_assessment_boxplot.py", + "apm_quality_assessment_export.py", + "apm_quality_assessment_gencfgs.py", + "apm_quality_assessment_optimize.py", + ] + outputs = [ "$root_build_dir/py_quality_assessment/{{source_file_part}}" ] + deps = [ + ":apm_configs", + ":lib", + ":output", + "../../../../resources/audio_processing/test/py_quality_assessment:probing_signals", + "../../../../rtc_tools:audioproc_f", + ] + } + + copy("apm_configs") { + testonly = true + sources = [ "apm_configs/default.json" ] + visibility = [ ":*" ] # Only targets in this file can depend on this. + outputs = [ + "$root_build_dir/py_quality_assessment/apm_configs/{{source_file_part}}", + ] + } # apm_configs + + copy("lib") { + testonly = true + sources = [ + "quality_assessment/__init__.py", + "quality_assessment/annotations.py", + "quality_assessment/audioproc_wrapper.py", + "quality_assessment/collect_data.py", + "quality_assessment/data_access.py", + "quality_assessment/echo_path_simulation.py", + "quality_assessment/echo_path_simulation_factory.py", + "quality_assessment/eval_scores.py", + "quality_assessment/eval_scores_factory.py", + "quality_assessment/evaluation.py", + "quality_assessment/exceptions.py", + "quality_assessment/export.py", + "quality_assessment/export_unittest.py", + "quality_assessment/external_vad.py", + "quality_assessment/input_mixer.py", + "quality_assessment/input_signal_creator.py", + "quality_assessment/results.css", + "quality_assessment/results.js", + "quality_assessment/signal_processing.py", + "quality_assessment/simulation.py", + "quality_assessment/test_data_generation.py", + "quality_assessment/test_data_generation_factory.py", + ] + visibility = [ ":*" ] # Only targets in this file can depend on this. + outputs = [ "$root_build_dir/py_quality_assessment/quality_assessment/{{source_file_part}}" ] + deps = [ "../../../../resources/audio_processing/test/py_quality_assessment:noise_tracks" ] + } + + copy("output") { + testonly = true + sources = [ "output/README.md" ] + visibility = [ ":*" ] # Only targets in this file can depend on this. + outputs = + [ "$root_build_dir/py_quality_assessment/output/{{source_file_part}}" ] + } + + group("unit_tests") { + testonly = true + visibility = [ ":*" ] # Only targets in this file can depend on this. + deps = [ + ":apm_vad", + ":fake_polqa", + ":lib_unit_tests", + ":scripts_unit_tests", + ":vad", + ] + } + + rtc_executable("fake_polqa") { + testonly = true + sources = [ "quality_assessment/fake_polqa.cc" ] + visibility = [ ":*" ] # Only targets in this file can depend on this. + output_dir = "${root_out_dir}/py_quality_assessment/quality_assessment" + deps = [ + "../../../../rtc_base:checks", + "//third_party/abseil-cpp/absl/strings", + ] + } + + rtc_executable("vad") { + testonly = true + sources = [ "quality_assessment/vad.cc" ] + deps = [ + "../../../../common_audio", + "../../../../rtc_base:logging", + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/flags:parse", + ] + } + + rtc_executable("apm_vad") { + testonly = true + sources = [ "quality_assessment/apm_vad.cc" ] + deps = [ + "../..", + "../../../../common_audio", + "../../../../rtc_base:logging", + "../../vad", + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/flags:parse", + ] + } + + rtc_executable("sound_level") { + testonly = true + sources = [ "quality_assessment/sound_level.cc" ] + deps = [ + "../..", + "../../../../common_audio", + "../../../../rtc_base:logging", + "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/abseil-cpp/absl/flags:parse", + ] + } + + copy("lib_unit_tests") { + testonly = true + sources = [ + "quality_assessment/annotations_unittest.py", + "quality_assessment/echo_path_simulation_unittest.py", + "quality_assessment/eval_scores_unittest.py", + "quality_assessment/fake_external_vad.py", + "quality_assessment/input_mixer_unittest.py", + "quality_assessment/signal_processing_unittest.py", + "quality_assessment/simulation_unittest.py", + "quality_assessment/test_data_generation_unittest.py", + ] + visibility = [ ":*" ] # Only targets in this file can depend on this. + outputs = [ "$root_build_dir/py_quality_assessment/quality_assessment/{{source_file_part}}" ] + } + + copy("scripts_unit_tests") { + testonly = true + sources = [ "apm_quality_assessment_unittest.py" ] + visibility = [ ":*" ] # Only targets in this file can depend on this. + outputs = [ "$root_build_dir/py_quality_assessment/{{source_file_part}}" ] + } +} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS new file mode 100644 index 0000000000..9f56bb830d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS @@ -0,0 +1,5 @@ +aleloi@webrtc.org +alessiob@webrtc.org +henrik.lundin@webrtc.org +ivoc@webrtc.org +peah@webrtc.org diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md new file mode 100644 index 0000000000..4156112df2 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md @@ -0,0 +1,125 @@ +# APM Quality Assessment tool + +Python wrapper of APM simulators (e.g., `audioproc_f`) with which quality +assessment can be automatized. The tool allows to simulate different noise +conditions, input signals, APM configurations and it computes different scores. +Once the scores are computed, the results can be easily exported to an HTML page +which allows to listen to the APM input and output signals and also the +reference one used for evaluation. + +## Dependencies + - OS: Linux + - Python 2.7 + - Python libraries: enum34, numpy, scipy, pydub (0.17.0+), pandas (0.20.1+), + pyquery (1.2+), jsmin (2.2+), csscompressor (0.9.4) + - It is recommended that a dedicated Python environment is used + - install `virtualenv` + - `$ sudo apt-get install python-virtualenv` + - setup a new Python environment (e.g., `my_env`) + - `$ cd ~ && virtualenv my_env` + - activate the new Python environment + - `$ source ~/my_env/bin/activate` + - add dependcies via `pip` + - `(my_env)$ pip install enum34 numpy pydub scipy pandas pyquery jsmin \` + `csscompressor` + - PolqaOem64 (see http://www.polqa.info/) + - Tested with POLQA Library v1.180 / P863 v2.400 + - Aachen Impulse Response (AIR) Database + - Download https://www2.iks.rwth-aachen.de/air/air_database_release_1_4.zip + - Input probing signals and noise tracks (you can make your own dataset - *1) + +## Build + - Compile WebRTC + - Go to `out/Default/py_quality_assessment` and check that + `apm_quality_assessment.py` exists + +## Unit tests + - Compile WebRTC + - Go to `out/Default/py_quality_assessment` + - Run `python -m unittest discover -p "*_unittest.py"` + +## First time setup + - Deploy PolqaOem64 and set the `POLQA_PATH` environment variable + - e.g., `$ export POLQA_PATH=/var/opt/PolqaOem64` + - Deploy the AIR Database and set the `AECHEN_IR_DATABASE_PATH` environment + variable + - e.g., `$ export AECHEN_IR_DATABASE_PATH=/var/opt/AIR_1_4` + - Deploy probing signal tracks into + - `out/Default/py_quality_assessment/probing_signals` (*1) + - Deploy noise tracks into + - `out/Default/py_quality_assessment/noise_tracks` (*1, *2) + +(*1) You can use custom files as long as they are mono tracks sampled at 48kHz +encoded in the 16 bit signed format (it is recommended that the tracks are +converted and exported with Audacity). + +## Usage (scores computation) + - Go to `out/Default/py_quality_assessment` + - Check the `apm_quality_assessment.sh` as an example script to parallelize the + experiments + - Adjust the script according to your preferences (e.g., output path) + - Run `apm_quality_assessment.sh` + - The script will end by opening the browser and showing ALL the computed + scores + +## Usage (export reports) +Showing all the results at once can be confusing. You therefore may want to +export separate reports. In this case, you can use the +`apm_quality_assessment_export.py` script as follows: + + - Set `--output_dir, -o` to the same value used in `apm_quality_assessment.sh` + - Use regular expressions to select/filter out scores by + - APM configurations: `--config_names, -c` + - capture signals: `--capture_names, -i` + - render signals: `--render_names, -r` + - echo simulator: `--echo_simulator_names, -e` + - test data generators: `--test_data_generators, -t` + - scores: `--eval_scores, -s` + - Assign a suffix to the report name using `-f ` + +For instance: + +``` +$ ./apm_quality_assessment_export.py \ + -o output/ \ + -c "(^default$)|(.*AE.*)" \ + -t \(white_noise\) \ + -s \(polqa\) \ + -f echo +``` + +## Usage (boxplot) +After generating stats, it can help to visualize how a score depends on a +certain APM simulator parameter. The `apm_quality_assessment_boxplot.py` script +helps with that, producing plots similar to [this +one](https://matplotlib.org/mpl_examples/pylab_examples/boxplot_demo_06.png). + +Suppose some scores come from running the APM simulator `audioproc_f` with +or without the level controller: `--lc=1` or `--lc=0`. Then two boxplots +side by side can be generated with + +``` +$ ./apm_quality_assessment_boxplot.py \ + -o /path/to/output + -v + -n /path/to/dir/with/apm_configs + -z lc +``` + +## Troubleshooting +The input wav file must be: + - sampled at a sample rate that is a multiple of 100 (required by POLQA) + - in the 16 bit format (required by `audioproc_f`) + - encoded in the Microsoft WAV signed 16 bit PCM format (Audacity default + when exporting) + +Depending on the license, the POLQA tool may take “breaks” as a way to limit the +throughput. When this happens, the APM Quality Assessment tool is slowed down. +For more details about this limitation, check Section 10.9.1 in the POLQA manual +v.1.18. + +In case of issues with the POLQA score computation, check +`py_quality_assessment/eval_scores.py` and adapt +`PolqaScore._parse_output_file()`. +The code can be also fixed directly into the build directory (namely, +`out/Default/py_quality_assessment/eval_scores.py`). diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json new file mode 100644 index 0000000000..5c3277bac0 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json @@ -0,0 +1 @@ +{"-all_default": null} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py new file mode 100755 index 0000000000..e067ecb692 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py @@ -0,0 +1,217 @@ +#!/usr/bin/env python +# Copyright (c) 2017 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. +"""Perform APM module quality assessment on one or more input files using one or + more APM simulator configuration files and one or more test data generators. + +Usage: apm_quality_assessment.py -i audio1.wav [audio2.wav ...] + -c cfg1.json [cfg2.json ...] + -n white [echo ...] + -e audio_level [polqa ...] + -o /path/to/output +""" + +import argparse +import logging +import os +import sys + +import quality_assessment.audioproc_wrapper as audioproc_wrapper +import quality_assessment.echo_path_simulation as echo_path_simulation +import quality_assessment.eval_scores as eval_scores +import quality_assessment.evaluation as evaluation +import quality_assessment.eval_scores_factory as eval_scores_factory +import quality_assessment.external_vad as external_vad +import quality_assessment.test_data_generation as test_data_generation +import quality_assessment.test_data_generation_factory as \ + test_data_generation_factory +import quality_assessment.simulation as simulation + +_ECHO_PATH_SIMULATOR_NAMES = ( + echo_path_simulation.EchoPathSimulator.REGISTERED_CLASSES) +_TEST_DATA_GENERATOR_CLASSES = ( + test_data_generation.TestDataGenerator.REGISTERED_CLASSES) +_TEST_DATA_GENERATORS_NAMES = _TEST_DATA_GENERATOR_CLASSES.keys() +_EVAL_SCORE_WORKER_CLASSES = eval_scores.EvaluationScore.REGISTERED_CLASSES +_EVAL_SCORE_WORKER_NAMES = _EVAL_SCORE_WORKER_CLASSES.keys() + +_DEFAULT_CONFIG_FILE = 'apm_configs/default.json' + +_POLQA_BIN_NAME = 'PolqaOem64' + + +def _InstanceArgumentsParser(): + """Arguments parser factory. + """ + parser = argparse.ArgumentParser(description=( + 'Perform APM module quality assessment on one or more input files using ' + 'one or more APM simulator configuration files and one or more ' + 'test data generators.')) + + parser.add_argument('-c', + '--config_files', + nargs='+', + required=False, + help=('path to the configuration files defining the ' + 'arguments with which the APM simulator tool is ' + 'called'), + default=[_DEFAULT_CONFIG_FILE]) + + parser.add_argument( + '-i', + '--capture_input_files', + nargs='+', + required=True, + help='path to the capture input wav files (one or more)') + + parser.add_argument('-r', + '--render_input_files', + nargs='+', + required=False, + help=('path to the render input wav files; either ' + 'omitted or one file for each file in ' + '--capture_input_files (files will be paired by ' + 'index)'), + default=None) + + parser.add_argument('-p', + '--echo_path_simulator', + required=False, + help=('custom echo path simulator name; required if ' + '--render_input_files is specified'), + choices=_ECHO_PATH_SIMULATOR_NAMES, + default=echo_path_simulation.NoEchoPathSimulator.NAME) + + parser.add_argument('-t', + '--test_data_generators', + nargs='+', + required=False, + help='custom list of test data generators to use', + choices=_TEST_DATA_GENERATORS_NAMES, + default=_TEST_DATA_GENERATORS_NAMES) + + parser.add_argument('--additive_noise_tracks_path', required=False, + help='path to the wav files for the additive', + default=test_data_generation. \ + AdditiveNoiseTestDataGenerator. \ + DEFAULT_NOISE_TRACKS_PATH) + + parser.add_argument('-e', + '--eval_scores', + nargs='+', + required=False, + help='custom list of evaluation scores to use', + choices=_EVAL_SCORE_WORKER_NAMES, + default=_EVAL_SCORE_WORKER_NAMES) + + parser.add_argument('-o', + '--output_dir', + required=False, + help=('base path to the output directory in which the ' + 'output wav files and the evaluation outcomes ' + 'are saved'), + default='output') + + parser.add_argument('--polqa_path', + required=True, + help='path to the POLQA tool') + + parser.add_argument('--air_db_path', + required=True, + help='path to the Aechen IR database') + + parser.add_argument('--apm_sim_path', required=False, + help='path to the APM simulator tool', + default=audioproc_wrapper. \ + AudioProcWrapper. \ + DEFAULT_APM_SIMULATOR_BIN_PATH) + + parser.add_argument('--echo_metric_tool_bin_path', + required=False, + help=('path to the echo metric binary ' + '(required for the echo eval score)'), + default=None) + + parser.add_argument( + '--copy_with_identity_generator', + required=False, + help=('If true, the identity test data generator makes a ' + 'copy of the clean speech input file.'), + default=False) + + parser.add_argument('--external_vad_paths', + nargs='+', + required=False, + help=('Paths to external VAD programs. Each must take' + '\'-i -o \' inputs'), + default=[]) + + parser.add_argument('--external_vad_names', + nargs='+', + required=False, + help=('Keys to the vad paths. Must be different and ' + 'as many as the paths.'), + default=[]) + + return parser + + +def _ValidateArguments(args, parser): + if args.capture_input_files and args.render_input_files and (len( + args.capture_input_files) != len(args.render_input_files)): + parser.error( + '--render_input_files and --capture_input_files must be lists ' + 'having the same length') + sys.exit(1) + + if args.render_input_files and not args.echo_path_simulator: + parser.error( + 'when --render_input_files is set, --echo_path_simulator is ' + 'also required') + sys.exit(1) + + if len(args.external_vad_names) != len(args.external_vad_paths): + parser.error('If provided, --external_vad_paths and ' + '--external_vad_names must ' + 'have the same number of arguments.') + sys.exit(1) + + +def main(): + # TODO(alessiob): level = logging.INFO once debugged. + logging.basicConfig(level=logging.DEBUG) + parser = _InstanceArgumentsParser() + args = parser.parse_args() + _ValidateArguments(args, parser) + + simulator = simulation.ApmModuleSimulator( + test_data_generator_factory=( + test_data_generation_factory.TestDataGeneratorFactory( + aechen_ir_database_path=args.air_db_path, + noise_tracks_path=args.additive_noise_tracks_path, + copy_with_identity=args.copy_with_identity_generator)), + evaluation_score_factory=eval_scores_factory. + EvaluationScoreWorkerFactory( + polqa_tool_bin_path=os.path.join(args.polqa_path, _POLQA_BIN_NAME), + echo_metric_tool_bin_path=args.echo_metric_tool_bin_path), + ap_wrapper=audioproc_wrapper.AudioProcWrapper(args.apm_sim_path), + evaluator=evaluation.ApmModuleEvaluator(), + external_vads=external_vad.ExternalVad.ConstructVadDict( + args.external_vad_paths, args.external_vad_names)) + simulator.Run(config_filepaths=args.config_files, + capture_input_filepaths=args.capture_input_files, + render_input_filepaths=args.render_input_files, + echo_path_simulator_name=args.echo_path_simulator, + test_data_generator_names=args.test_data_generators, + eval_score_names=args.eval_scores, + output_dir=args.output_dir) + sys.exit(0) + + +if __name__ == '__main__': + main() diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh new file mode 100755 index 0000000000..aa563ee26b --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh @@ -0,0 +1,91 @@ +#!/bin/bash +# Copyright (c) 2017 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. + +# Path to the POLQA tool. +if [ -z ${POLQA_PATH} ]; then # Check if defined. + # Default location. + export POLQA_PATH='/var/opt/PolqaOem64' +fi +if [ -d "${POLQA_PATH}" ]; then + echo "POLQA found in ${POLQA_PATH}" +else + echo "POLQA not found in ${POLQA_PATH}" + exit 1 +fi + +# Path to the Aechen IR database. +if [ -z ${AECHEN_IR_DATABASE_PATH} ]; then # Check if defined. + # Default location. + export AECHEN_IR_DATABASE_PATH='/var/opt/AIR_1_4' +fi +if [ -d "${AECHEN_IR_DATABASE_PATH}" ]; then + echo "AIR database found in ${AECHEN_IR_DATABASE_PATH}" +else + echo "AIR database not found in ${AECHEN_IR_DATABASE_PATH}" + exit 1 +fi + +# Customize probing signals, test data generators and scores if needed. +CAPTURE_SIGNALS=(probing_signals/*.wav) +TEST_DATA_GENERATORS=( \ + "identity" \ + "white_noise" \ + # "environmental_noise" \ + # "reverberation" \ +) +SCORES=( \ + # "polqa" \ + "audio_level_peak" \ + "audio_level_mean" \ +) +OUTPUT_PATH=output + +# Generate standard APM config files. +chmod +x apm_quality_assessment_gencfgs.py +./apm_quality_assessment_gencfgs.py + +# Customize APM configurations if needed. +APM_CONFIGS=(apm_configs/*.json) + +# Add output path if missing. +if [ ! -d ${OUTPUT_PATH} ]; then + mkdir ${OUTPUT_PATH} +fi + +# Start one process for each "probing signal"-"test data source" pair. +chmod +x apm_quality_assessment.py +for capture_signal_filepath in "${CAPTURE_SIGNALS[@]}" ; do + probing_signal_name="$(basename $capture_signal_filepath)" + probing_signal_name="${probing_signal_name%.*}" + for test_data_gen_name in "${TEST_DATA_GENERATORS[@]}" ; do + LOG_FILE="${OUTPUT_PATH}/apm_qa-${probing_signal_name}-"` + `"${test_data_gen_name}.log" + echo "Starting ${probing_signal_name} ${test_data_gen_name} "` + `"(see ${LOG_FILE})" + ./apm_quality_assessment.py \ + --polqa_path ${POLQA_PATH}\ + --air_db_path ${AECHEN_IR_DATABASE_PATH}\ + -i ${capture_signal_filepath} \ + -o ${OUTPUT_PATH} \ + -t ${test_data_gen_name} \ + -c "${APM_CONFIGS[@]}" \ + -e "${SCORES[@]}" > $LOG_FILE 2>&1 & + done +done + +# Join Python processes running apm_quality_assessment.py. +wait + +# Export results. +chmod +x ./apm_quality_assessment_export.py +./apm_quality_assessment_export.py -o ${OUTPUT_PATH} + +# Show results in the browser. +RESULTS_FILE="$(realpath ${OUTPUT_PATH}/results.html)" +sensible-browser "file://${RESULTS_FILE}" > /dev/null 2>&1 & diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py new file mode 100644 index 0000000000..c425885b95 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py @@ -0,0 +1,154 @@ +#!/usr/bin/env python +# Copyright (c) 2017 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. +"""Shows boxplots of given score for different values of selected +parameters. Can be used to compare scores by audioproc_f flag. + +Usage: apm_quality_assessment_boxplot.py -o /path/to/output + -v polqa + -n /path/to/dir/with/apm_configs + -z audioproc_f_arg1 [arg2 ...] + +Arguments --config_names, --render_names, --echo_simulator_names, +--test_data_generators, --eval_scores can be used to filter the data +used for plotting. +""" + +import collections +import logging +import matplotlib.pyplot as plt +import os + +import quality_assessment.data_access as data_access +import quality_assessment.collect_data as collect_data + + +def InstanceArgumentsParser(): + """Arguments parser factory. + """ + parser = collect_data.InstanceArgumentsParser() + parser.description = ( + 'Shows boxplot of given score for different values of selected' + 'parameters. Can be used to compare scores by audioproc_f flag') + + parser.add_argument('-v', + '--eval_score', + required=True, + help=('Score name for constructing boxplots')) + + parser.add_argument( + '-n', + '--config_dir', + required=False, + help=('path to the folder with the configuration files'), + default='apm_configs') + + parser.add_argument('-z', + '--params_to_plot', + required=True, + nargs='+', + help=('audioproc_f parameter values' + 'by which to group scores (no leading dash)')) + + return parser + + +def FilterScoresByParams(data_frame, filter_params, score_name, config_dir): + """Filters data on the values of one or more parameters. + + Args: + data_frame: pandas.DataFrame of all used input data. + + filter_params: each config of the input data is assumed to have + exactly one parameter from `filter_params` defined. Every value + of the parameters in `filter_params` is a key in the returned + dict; the associated value is all cells of the data with that + value of the parameter. + + score_name: Name of score which value is boxplotted. Currently cannot do + more than one value. + + config_dir: path to dir with APM configs. + + Returns: dictionary, key is a param value, result is all scores for + that param value (see `filter_params` for explanation). + """ + results = collections.defaultdict(dict) + config_names = data_frame['apm_config'].drop_duplicates().values.tolist() + + for config_name in config_names: + config_json = data_access.AudioProcConfigFile.Load( + os.path.join(config_dir, config_name + '.json')) + data_with_config = data_frame[data_frame.apm_config == config_name] + data_cell_scores = data_with_config[data_with_config.eval_score_name == + score_name] + + # Exactly one of `params_to_plot` must match: + (matching_param, ) = [ + x for x in filter_params if '-' + x in config_json + ] + + # Add scores for every track to the result. + for capture_name in data_cell_scores.capture: + result_score = float(data_cell_scores[data_cell_scores.capture == + capture_name].score) + config_dict = results[config_json['-' + matching_param]] + if capture_name not in config_dict: + config_dict[capture_name] = {} + + config_dict[capture_name][matching_param] = result_score + + return results + + +def _FlattenToScoresList(config_param_score_dict): + """Extracts a list of scores from input data structure. + + Args: + config_param_score_dict: of the form {'capture_name': + {'param_name' : score_value,.. } ..} + + Returns: Plain list of all score value present in input data + structure + """ + result = [] + for capture_name in config_param_score_dict: + result += list(config_param_score_dict[capture_name].values()) + return result + + +def main(): + # Init. + # TODO(alessiob): INFO once debugged. + logging.basicConfig(level=logging.DEBUG) + parser = InstanceArgumentsParser() + args = parser.parse_args() + + # Get the scores. + src_path = collect_data.ConstructSrcPath(args) + logging.debug(src_path) + scores_data_frame = collect_data.FindScores(src_path, args) + + # Filter the data by `args.params_to_plot` + scores_filtered = FilterScoresByParams(scores_data_frame, + args.params_to_plot, + args.eval_score, args.config_dir) + + data_list = sorted(scores_filtered.items()) + data_values = [_FlattenToScoresList(x) for (_, x) in data_list] + data_labels = [x for (x, _) in data_list] + + _, axes = plt.subplots(nrows=1, ncols=1, figsize=(6, 6)) + axes.boxplot(data_values, labels=data_labels) + axes.set_ylabel(args.eval_score) + axes.set_xlabel('/'.join(args.params_to_plot)) + plt.show() + + +if __name__ == "__main__": + main() diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py new file mode 100755 index 0000000000..c20accb9dc --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py @@ -0,0 +1,63 @@ +#!/usr/bin/env python +# Copyright (c) 2017 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. +"""Export the scores computed by the apm_quality_assessment.py script into an + HTML file. +""" + +import logging +import os +import sys + +import quality_assessment.collect_data as collect_data +import quality_assessment.export as export + + +def _BuildOutputFilename(filename_suffix): + """Builds the filename for the exported file. + + Args: + filename_suffix: suffix for the output file name. + + Returns: + A string. + """ + if filename_suffix is None: + return 'results.html' + return 'results-{}.html'.format(filename_suffix) + + +def main(): + # Init. + logging.basicConfig( + level=logging.DEBUG) # TODO(alessio): INFO once debugged. + parser = collect_data.InstanceArgumentsParser() + parser.add_argument('-f', + '--filename_suffix', + help=('suffix of the exported file')) + parser.description = ('Exports pre-computed APM module quality assessment ' + 'results into HTML tables') + args = parser.parse_args() + + # Get the scores. + src_path = collect_data.ConstructSrcPath(args) + logging.debug(src_path) + scores_data_frame = collect_data.FindScores(src_path, args) + + # Export. + output_filepath = os.path.join(args.output_dir, + _BuildOutputFilename(args.filename_suffix)) + exporter = export.HtmlExport(output_filepath) + exporter.Export(scores_data_frame) + + logging.info('output file successfully written in %s', output_filepath) + sys.exit(0) + + +if __name__ == '__main__': + main() diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py new file mode 100755 index 0000000000..ca80f85bd1 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py @@ -0,0 +1,128 @@ +#!/usr/bin/env python +# Copyright (c) 2017 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. +"""Generate .json files with which the APM module can be tested using the + apm_quality_assessment.py script and audioproc_f as APM simulator. +""" + +import logging +import os + +import quality_assessment.data_access as data_access + +OUTPUT_PATH = os.path.abspath('apm_configs') + + +def _GenerateDefaultOverridden(config_override): + """Generates one or more APM overriden configurations. + + For each item in config_override, it overrides the default configuration and + writes a new APM configuration file. + + The default settings are loaded via "-all_default". + Check "src/modules/audio_processing/test/audioproc_float.cc" and search + for "if (FLAG_all_default) {". + + For instance, in 55eb6d621489730084927868fed195d3645a9ec9 the default is this: + settings.use_aec = rtc::Optional(true); + settings.use_aecm = rtc::Optional(false); + settings.use_agc = rtc::Optional(true); + settings.use_bf = rtc::Optional(false); + settings.use_ed = rtc::Optional(false); + settings.use_hpf = rtc::Optional(true); + settings.use_le = rtc::Optional(true); + settings.use_ns = rtc::Optional(true); + settings.use_ts = rtc::Optional(true); + settings.use_vad = rtc::Optional(true); + + Args: + config_override: dict of APM configuration file names as keys; the values + are dict instances encoding the audioproc_f flags. + """ + for config_filename in config_override: + config = config_override[config_filename] + config['-all_default'] = None + + config_filepath = os.path.join( + OUTPUT_PATH, 'default-{}.json'.format(config_filename)) + logging.debug('config file <%s> | %s', config_filepath, config) + + data_access.AudioProcConfigFile.Save(config_filepath, config) + logging.info('config file created: <%s>', config_filepath) + + +def _GenerateAllDefaultButOne(): + """Disables the flags enabled by default one-by-one. + """ + config_sets = { + 'no_AEC': { + '-aec': 0, + }, + 'no_AGC': { + '-agc': 0, + }, + 'no_HP_filter': { + '-hpf': 0, + }, + 'no_level_estimator': { + '-le': 0, + }, + 'no_noise_suppressor': { + '-ns': 0, + }, + 'no_transient_suppressor': { + '-ts': 0, + }, + 'no_vad': { + '-vad': 0, + }, + } + _GenerateDefaultOverridden(config_sets) + + +def _GenerateAllDefaultPlusOne(): + """Enables the flags disabled by default one-by-one. + """ + config_sets = { + 'with_AECM': { + '-aec': 0, + '-aecm': 1, + }, # AEC and AECM are exclusive. + 'with_AGC_limiter': { + '-agc_limiter': 1, + }, + 'with_AEC_delay_agnostic': { + '-delay_agnostic': 1, + }, + 'with_drift_compensation': { + '-drift_compensation': 1, + }, + 'with_residual_echo_detector': { + '-ed': 1, + }, + 'with_AEC_extended_filter': { + '-extended_filter': 1, + }, + 'with_LC': { + '-lc': 1, + }, + 'with_refined_adaptive_filter': { + '-refined_adaptive_filter': 1, + }, + } + _GenerateDefaultOverridden(config_sets) + + +def main(): + logging.basicConfig(level=logging.INFO) + _GenerateAllDefaultPlusOne() + _GenerateAllDefaultButOne() + + +if __name__ == '__main__': + main() diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py new file mode 100644 index 0000000000..ecae2ed995 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py @@ -0,0 +1,189 @@ +#!/usr/bin/env python +# Copyright (c) 2017 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. +"""Finds the APM configuration that maximizes a provided metric by +parsing the output generated apm_quality_assessment.py. +""" + +from __future__ import division + +import collections +import logging +import os + +import quality_assessment.data_access as data_access +import quality_assessment.collect_data as collect_data + + +def _InstanceArgumentsParser(): + """Arguments parser factory. Extends the arguments from 'collect_data' + with a few extra for selecting what parameters to optimize for. + """ + parser = collect_data.InstanceArgumentsParser() + parser.description = ( + 'Rudimentary optimization of a function over different parameter' + 'combinations.') + + parser.add_argument( + '-n', + '--config_dir', + required=False, + help=('path to the folder with the configuration files'), + default='apm_configs') + + parser.add_argument('-p', + '--params', + required=True, + nargs='+', + help=('parameters to parse from the config files in' + 'config_dir')) + + parser.add_argument( + '-z', + '--params_not_to_optimize', + required=False, + nargs='+', + default=[], + help=('parameters from `params` not to be optimized for')) + + return parser + + +def _ConfigurationAndScores(data_frame, params, params_not_to_optimize, + config_dir): + """Returns a list of all configurations and scores. + + Args: + data_frame: A pandas data frame with the scores and config name + returned by _FindScores. + params: The parameter names to parse from configs the config + directory + + params_not_to_optimize: The parameter names which shouldn't affect + the optimal parameter + selection. E.g., fixed settings and not + tunable parameters. + + config_dir: Path to folder with config files. + + Returns: + Dictionary of the form + {param_combination: [{params: {param1: value1, ...}, + scores: {score1: value1, ...}}]}. + + The key `param_combination` runs over all parameter combinations + of the parameters in `params` and not in + `params_not_to_optimize`. A corresponding value is a list of all + param combinations for params in `params_not_to_optimize` and + their scores. + """ + results = collections.defaultdict(list) + config_names = data_frame['apm_config'].drop_duplicates().values.tolist() + score_names = data_frame['eval_score_name'].drop_duplicates( + ).values.tolist() + + # Normalize the scores + normalization_constants = {} + for score_name in score_names: + scores = data_frame[data_frame.eval_score_name == score_name].score + normalization_constants[score_name] = max(scores) + + params_to_optimize = [p for p in params if p not in params_not_to_optimize] + param_combination = collections.namedtuple("ParamCombination", + params_to_optimize) + + for config_name in config_names: + config_json = data_access.AudioProcConfigFile.Load( + os.path.join(config_dir, config_name + ".json")) + scores = {} + data_cell = data_frame[data_frame.apm_config == config_name] + for score_name in score_names: + data_cell_scores = data_cell[data_cell.eval_score_name == + score_name].score + scores[score_name] = sum(data_cell_scores) / len(data_cell_scores) + scores[score_name] /= normalization_constants[score_name] + + result = {'scores': scores, 'params': {}} + config_optimize_params = {} + for param in params: + if param in params_to_optimize: + config_optimize_params[param] = config_json['-' + param] + else: + result['params'][param] = config_json['-' + param] + + current_param_combination = param_combination(**config_optimize_params) + results[current_param_combination].append(result) + return results + + +def _FindOptimalParameter(configs_and_scores, score_weighting): + """Finds the config producing the maximal score. + + Args: + configs_and_scores: structure of the form returned by + _ConfigurationAndScores + + score_weighting: a function to weight together all score values of + the form [{params: {param1: value1, ...}, scores: + {score1: value1, ...}}] into a numeric + value + Returns: + the config that has the largest values of `score_weighting` applied + to its scores. + """ + + min_score = float('+inf') + best_params = None + for config in configs_and_scores: + scores_and_params = configs_and_scores[config] + current_score = score_weighting(scores_and_params) + if current_score < min_score: + min_score = current_score + best_params = config + logging.debug("Score: %f", current_score) + logging.debug("Config: %s", str(config)) + return best_params + + +def _ExampleWeighting(scores_and_configs): + """Example argument to `_FindOptimalParameter` + Args: + scores_and_configs: a list of configs and scores, in the form + described in _FindOptimalParameter + Returns: + numeric value, the sum of all scores + """ + res = 0 + for score_config in scores_and_configs: + res += sum(score_config['scores'].values()) + return res + + +def main(): + # Init. + # TODO(alessiob): INFO once debugged. + logging.basicConfig(level=logging.DEBUG) + parser = _InstanceArgumentsParser() + args = parser.parse_args() + + # Get the scores. + src_path = collect_data.ConstructSrcPath(args) + logging.debug('Src path <%s>', src_path) + scores_data_frame = collect_data.FindScores(src_path, args) + all_scores = _ConfigurationAndScores(scores_data_frame, args.params, + args.params_not_to_optimize, + args.config_dir) + + opt_param = _FindOptimalParameter(all_scores, _ExampleWeighting) + + logging.info('Optimal parameter combination: <%s>', opt_param) + logging.info('It\'s score values: <%s>', all_scores[opt_param]) + + +if __name__ == "__main__": + main() diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py new file mode 100644 index 0000000000..80338c1373 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py @@ -0,0 +1,28 @@ +# Copyright (c) 2017 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. +"""Unit tests for the apm_quality_assessment module. +""" + +import sys +import unittest + +import mock + +import apm_quality_assessment + + +class TestSimulationScript(unittest.TestCase): + """Unit tests for the apm_quality_assessment module. + """ + + def testMain(self): + # Exit with error code if no arguments are passed. + with self.assertRaises(SystemExit) as cm, mock.patch.object( + sys, 'argv', ['apm_quality_assessment.py']): + apm_quality_assessment.main() + self.assertGreater(cm.exception.code, 0) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md new file mode 100644 index 0000000000..66e2a1c848 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md @@ -0,0 +1 @@ +You can use this folder for the output generated by the apm_quality_assessment scripts. diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py new file mode 100644 index 0000000000..b870dfaef3 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py @@ -0,0 +1,7 @@ +# Copyright (c) 2017 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. diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py new file mode 100644 index 0000000000..93a8248397 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py @@ -0,0 +1,296 @@ +# Copyright (c) 2017 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. +"""Extraction of annotations from audio files. +""" + +from __future__ import division +import logging +import os +import shutil +import struct +import subprocess +import sys +import tempfile + +try: + import numpy as np +except ImportError: + logging.critical('Cannot import the third-party Python package numpy') + sys.exit(1) + +from . import external_vad +from . import exceptions +from . import signal_processing + + +class AudioAnnotationsExtractor(object): + """Extracts annotations from audio files. + """ + + class VadType(object): + ENERGY_THRESHOLD = 1 # TODO(alessiob): Consider switching to P56 standard. + WEBRTC_COMMON_AUDIO = 2 # common_audio/vad/include/vad.h + WEBRTC_APM = 4 # modules/audio_processing/vad/vad.h + + def __init__(self, value): + if (not isinstance(value, int)) or not 0 <= value <= 7: + raise exceptions.InitializationException('Invalid vad type: ' + + value) + self._value = value + + def Contains(self, vad_type): + return self._value | vad_type == self._value + + def __str__(self): + vads = [] + if self.Contains(self.ENERGY_THRESHOLD): + vads.append("energy") + if self.Contains(self.WEBRTC_COMMON_AUDIO): + vads.append("common_audio") + if self.Contains(self.WEBRTC_APM): + vads.append("apm") + return "VadType({})".format(", ".join(vads)) + + _OUTPUT_FILENAME_TEMPLATE = '{}annotations.npz' + + # Level estimation params. + _ONE_DB_REDUCTION = np.power(10.0, -1.0 / 20.0) + _LEVEL_FRAME_SIZE_MS = 1.0 + # The time constants in ms indicate the time it takes for the level estimate + # to go down/up by 1 db if the signal is zero. + _LEVEL_ATTACK_MS = 5.0 + _LEVEL_DECAY_MS = 20.0 + + # VAD params. + _VAD_THRESHOLD = 1 + _VAD_WEBRTC_PATH = os.path.join(os.path.dirname(os.path.abspath(__file__)), + os.pardir, os.pardir) + _VAD_WEBRTC_COMMON_AUDIO_PATH = os.path.join(_VAD_WEBRTC_PATH, 'vad') + + _VAD_WEBRTC_APM_PATH = os.path.join(_VAD_WEBRTC_PATH, 'apm_vad') + + def __init__(self, vad_type, external_vads=None): + self._signal = None + self._level = None + self._level_frame_size = None + self._common_audio_vad = None + self._energy_vad = None + self._apm_vad_probs = None + self._apm_vad_rms = None + self._vad_frame_size = None + self._vad_frame_size_ms = None + self._c_attack = None + self._c_decay = None + + self._vad_type = self.VadType(vad_type) + logging.info('VADs used for annotations: ' + str(self._vad_type)) + + if external_vads is None: + external_vads = {} + self._external_vads = external_vads + + assert len(self._external_vads) == len(external_vads), ( + 'The external VAD names must be unique.') + for vad in external_vads.values(): + if not isinstance(vad, external_vad.ExternalVad): + raise exceptions.InitializationException('Invalid vad type: ' + + str(type(vad))) + logging.info('External VAD used for annotation: ' + str(vad.name)) + + assert os.path.exists(self._VAD_WEBRTC_COMMON_AUDIO_PATH), \ + self._VAD_WEBRTC_COMMON_AUDIO_PATH + assert os.path.exists(self._VAD_WEBRTC_APM_PATH), \ + self._VAD_WEBRTC_APM_PATH + + @classmethod + def GetOutputFileNameTemplate(cls): + return cls._OUTPUT_FILENAME_TEMPLATE + + def GetLevel(self): + return self._level + + def GetLevelFrameSize(self): + return self._level_frame_size + + @classmethod + def GetLevelFrameSizeMs(cls): + return cls._LEVEL_FRAME_SIZE_MS + + def GetVadOutput(self, vad_type): + if vad_type == self.VadType.ENERGY_THRESHOLD: + return self._energy_vad + elif vad_type == self.VadType.WEBRTC_COMMON_AUDIO: + return self._common_audio_vad + elif vad_type == self.VadType.WEBRTC_APM: + return (self._apm_vad_probs, self._apm_vad_rms) + else: + raise exceptions.InitializationException('Invalid vad type: ' + + vad_type) + + def GetVadFrameSize(self): + return self._vad_frame_size + + def GetVadFrameSizeMs(self): + return self._vad_frame_size_ms + + def Extract(self, filepath): + # Load signal. + self._signal = signal_processing.SignalProcessingUtils.LoadWav( + filepath) + if self._signal.channels != 1: + raise NotImplementedError( + 'Multiple-channel annotations not implemented') + + # Level estimation params. + self._level_frame_size = int(self._signal.frame_rate / 1000 * + (self._LEVEL_FRAME_SIZE_MS)) + self._c_attack = 0.0 if self._LEVEL_ATTACK_MS == 0 else ( + self._ONE_DB_REDUCTION**(self._LEVEL_FRAME_SIZE_MS / + self._LEVEL_ATTACK_MS)) + self._c_decay = 0.0 if self._LEVEL_DECAY_MS == 0 else ( + self._ONE_DB_REDUCTION**(self._LEVEL_FRAME_SIZE_MS / + self._LEVEL_DECAY_MS)) + + # Compute level. + self._LevelEstimation() + + # Ideal VAD output, it requires clean speech with high SNR as input. + if self._vad_type.Contains(self.VadType.ENERGY_THRESHOLD): + # Naive VAD based on level thresholding. + vad_threshold = np.percentile(self._level, self._VAD_THRESHOLD) + self._energy_vad = np.uint8(self._level > vad_threshold) + self._vad_frame_size = self._level_frame_size + self._vad_frame_size_ms = self._LEVEL_FRAME_SIZE_MS + if self._vad_type.Contains(self.VadType.WEBRTC_COMMON_AUDIO): + # WebRTC common_audio/ VAD. + self._RunWebRtcCommonAudioVad(filepath, self._signal.frame_rate) + if self._vad_type.Contains(self.VadType.WEBRTC_APM): + # WebRTC modules/audio_processing/ VAD. + self._RunWebRtcApmVad(filepath) + for extvad_name in self._external_vads: + self._external_vads[extvad_name].Run(filepath) + + def Save(self, output_path, annotation_name=""): + ext_kwargs = { + 'extvad_conf-' + ext_vad: + self._external_vads[ext_vad].GetVadOutput() + for ext_vad in self._external_vads + } + np.savez_compressed(file=os.path.join( + output_path, + self.GetOutputFileNameTemplate().format(annotation_name)), + level=self._level, + level_frame_size=self._level_frame_size, + level_frame_size_ms=self._LEVEL_FRAME_SIZE_MS, + vad_output=self._common_audio_vad, + vad_energy_output=self._energy_vad, + vad_frame_size=self._vad_frame_size, + vad_frame_size_ms=self._vad_frame_size_ms, + vad_probs=self._apm_vad_probs, + vad_rms=self._apm_vad_rms, + **ext_kwargs) + + def _LevelEstimation(self): + # Read samples. + samples = signal_processing.SignalProcessingUtils.AudioSegmentToRawData( + self._signal).astype(np.float32) / 32768.0 + num_frames = len(samples) // self._level_frame_size + num_samples = num_frames * self._level_frame_size + + # Envelope. + self._level = np.max(np.reshape(np.abs(samples[:num_samples]), + (num_frames, self._level_frame_size)), + axis=1) + assert len(self._level) == num_frames + + # Envelope smoothing. + smooth = lambda curr, prev, k: (1 - k) * curr + k * prev + self._level[0] = smooth(self._level[0], 0.0, self._c_attack) + for i in range(1, num_frames): + self._level[i] = smooth( + self._level[i], self._level[i - 1], self._c_attack if + (self._level[i] > self._level[i - 1]) else self._c_decay) + + def _RunWebRtcCommonAudioVad(self, wav_file_path, sample_rate): + self._common_audio_vad = None + self._vad_frame_size = None + + # Create temporary output path. + tmp_path = tempfile.mkdtemp() + output_file_path = os.path.join( + tmp_path, + os.path.split(wav_file_path)[1] + '_vad.tmp') + + # Call WebRTC VAD. + try: + subprocess.call([ + self._VAD_WEBRTC_COMMON_AUDIO_PATH, '-i', wav_file_path, '-o', + output_file_path + ], + cwd=self._VAD_WEBRTC_PATH) + + # Read bytes. + with open(output_file_path, 'rb') as f: + raw_data = f.read() + + # Parse side information. + self._vad_frame_size_ms = struct.unpack('B', raw_data[0])[0] + self._vad_frame_size = self._vad_frame_size_ms * sample_rate / 1000 + assert self._vad_frame_size_ms in [10, 20, 30] + extra_bits = struct.unpack('B', raw_data[-1])[0] + assert 0 <= extra_bits <= 8 + + # Init VAD vector. + num_bytes = len(raw_data) + num_frames = 8 * (num_bytes - + 2) - extra_bits # 8 frames for each byte. + self._common_audio_vad = np.zeros(num_frames, np.uint8) + + # Read VAD decisions. + for i, byte in enumerate(raw_data[1:-1]): + byte = struct.unpack('B', byte)[0] + for j in range(8 if i < num_bytes - 3 else (8 - extra_bits)): + self._common_audio_vad[i * 8 + j] = int(byte & 1) + byte = byte >> 1 + except Exception as e: + logging.error('Error while running the WebRTC VAD (' + e.message + + ')') + finally: + if os.path.exists(tmp_path): + shutil.rmtree(tmp_path) + + def _RunWebRtcApmVad(self, wav_file_path): + # Create temporary output path. + tmp_path = tempfile.mkdtemp() + output_file_path_probs = os.path.join( + tmp_path, + os.path.split(wav_file_path)[1] + '_vad_probs.tmp') + output_file_path_rms = os.path.join( + tmp_path, + os.path.split(wav_file_path)[1] + '_vad_rms.tmp') + + # Call WebRTC VAD. + try: + subprocess.call([ + self._VAD_WEBRTC_APM_PATH, '-i', wav_file_path, '-o_probs', + output_file_path_probs, '-o_rms', output_file_path_rms + ], + cwd=self._VAD_WEBRTC_PATH) + + # Parse annotations. + self._apm_vad_probs = np.fromfile(output_file_path_probs, + np.double) + self._apm_vad_rms = np.fromfile(output_file_path_rms, np.double) + assert len(self._apm_vad_rms) == len(self._apm_vad_probs) + + except Exception as e: + logging.error('Error while running the WebRTC APM VAD (' + + e.message + ')') + finally: + if os.path.exists(tmp_path): + shutil.rmtree(tmp_path) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py new file mode 100644 index 0000000000..8230208808 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py @@ -0,0 +1,160 @@ +# Copyright (c) 2017 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. +"""Unit tests for the annotations module. +""" + +from __future__ import division +import logging +import os +import shutil +import tempfile +import unittest + +import numpy as np + +from . import annotations +from . import external_vad +from . import input_signal_creator +from . import signal_processing + + +class TestAnnotationsExtraction(unittest.TestCase): + """Unit tests for the annotations module. + """ + + _CLEAN_TMP_OUTPUT = True + _DEBUG_PLOT_VAD = False + _VAD_TYPE_CLASS = annotations.AudioAnnotationsExtractor.VadType + _ALL_VAD_TYPES = (_VAD_TYPE_CLASS.ENERGY_THRESHOLD + | _VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO + | _VAD_TYPE_CLASS.WEBRTC_APM) + + def setUp(self): + """Create temporary folder.""" + self._tmp_path = tempfile.mkdtemp() + self._wav_file_path = os.path.join(self._tmp_path, 'tone.wav') + pure_tone, _ = input_signal_creator.InputSignalCreator.Create( + 'pure_tone', [440, 1000]) + signal_processing.SignalProcessingUtils.SaveWav( + self._wav_file_path, pure_tone) + self._sample_rate = pure_tone.frame_rate + + def tearDown(self): + """Recursively delete temporary folder.""" + if self._CLEAN_TMP_OUTPUT: + shutil.rmtree(self._tmp_path) + else: + logging.warning(self.id() + ' did not clean the temporary path ' + + (self._tmp_path)) + + def testFrameSizes(self): + e = annotations.AudioAnnotationsExtractor(self._ALL_VAD_TYPES) + e.Extract(self._wav_file_path) + samples_to_ms = lambda n, sr: 1000 * n // sr + self.assertEqual( + samples_to_ms(e.GetLevelFrameSize(), self._sample_rate), + e.GetLevelFrameSizeMs()) + self.assertEqual(samples_to_ms(e.GetVadFrameSize(), self._sample_rate), + e.GetVadFrameSizeMs()) + + def testVoiceActivityDetectors(self): + for vad_type_value in range(0, self._ALL_VAD_TYPES + 1): + vad_type = self._VAD_TYPE_CLASS(vad_type_value) + e = annotations.AudioAnnotationsExtractor(vad_type=vad_type_value) + e.Extract(self._wav_file_path) + if vad_type.Contains(self._VAD_TYPE_CLASS.ENERGY_THRESHOLD): + # pylint: disable=unpacking-non-sequence + vad_output = e.GetVadOutput( + self._VAD_TYPE_CLASS.ENERGY_THRESHOLD) + self.assertGreater(len(vad_output), 0) + self.assertGreaterEqual( + float(np.sum(vad_output)) / len(vad_output), 0.95) + + if vad_type.Contains(self._VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO): + # pylint: disable=unpacking-non-sequence + vad_output = e.GetVadOutput( + self._VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO) + self.assertGreater(len(vad_output), 0) + self.assertGreaterEqual( + float(np.sum(vad_output)) / len(vad_output), 0.95) + + if vad_type.Contains(self._VAD_TYPE_CLASS.WEBRTC_APM): + # pylint: disable=unpacking-non-sequence + (vad_probs, + vad_rms) = e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_APM) + self.assertGreater(len(vad_probs), 0) + self.assertGreater(len(vad_rms), 0) + self.assertGreaterEqual( + float(np.sum(vad_probs)) / len(vad_probs), 0.5) + self.assertGreaterEqual( + float(np.sum(vad_rms)) / len(vad_rms), 20000) + + if self._DEBUG_PLOT_VAD: + frame_times_s = lambda num_frames, frame_size_ms: np.arange( + num_frames).astype(np.float32) * frame_size_ms / 1000.0 + level = e.GetLevel() + t_level = frame_times_s(num_frames=len(level), + frame_size_ms=e.GetLevelFrameSizeMs()) + t_vad = frame_times_s(num_frames=len(vad_output), + frame_size_ms=e.GetVadFrameSizeMs()) + import matplotlib.pyplot as plt + plt.figure() + plt.hold(True) + plt.plot(t_level, level) + plt.plot(t_vad, vad_output * np.max(level), '.') + plt.show() + + def testSaveLoad(self): + e = annotations.AudioAnnotationsExtractor(self._ALL_VAD_TYPES) + e.Extract(self._wav_file_path) + e.Save(self._tmp_path, "fake-annotation") + + data = np.load( + os.path.join( + self._tmp_path, + e.GetOutputFileNameTemplate().format("fake-annotation"))) + np.testing.assert_array_equal(e.GetLevel(), data['level']) + self.assertEqual(np.float32, data['level'].dtype) + np.testing.assert_array_equal( + e.GetVadOutput(self._VAD_TYPE_CLASS.ENERGY_THRESHOLD), + data['vad_energy_output']) + np.testing.assert_array_equal( + e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO), + data['vad_output']) + np.testing.assert_array_equal( + e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_APM)[0], + data['vad_probs']) + np.testing.assert_array_equal( + e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_APM)[1], + data['vad_rms']) + self.assertEqual(np.uint8, data['vad_energy_output'].dtype) + self.assertEqual(np.float64, data['vad_probs'].dtype) + self.assertEqual(np.float64, data['vad_rms'].dtype) + + def testEmptyExternalShouldNotCrash(self): + for vad_type_value in range(0, self._ALL_VAD_TYPES + 1): + annotations.AudioAnnotationsExtractor(vad_type_value, {}) + + def testFakeExternalSaveLoad(self): + def FakeExternalFactory(): + return external_vad.ExternalVad( + os.path.join(os.path.dirname(os.path.abspath(__file__)), + 'fake_external_vad.py'), 'fake') + + for vad_type_value in range(0, self._ALL_VAD_TYPES + 1): + e = annotations.AudioAnnotationsExtractor( + vad_type_value, {'fake': FakeExternalFactory()}) + e.Extract(self._wav_file_path) + e.Save(self._tmp_path, annotation_name="fake-annotation") + data = np.load( + os.path.join( + self._tmp_path, + e.GetOutputFileNameTemplate().format("fake-annotation"))) + self.assertEqual(np.float32, data['extvad_conf-fake'].dtype) + np.testing.assert_almost_equal(np.arange(100, dtype=np.float32), + data['extvad_conf-fake']) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json new file mode 100644 index 0000000000..5c3277bac0 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json @@ -0,0 +1 @@ +{"-all_default": null} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc new file mode 100644 index 0000000000..73ce4ed3f7 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc @@ -0,0 +1,96 @@ +// Copyright (c) 2017 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 +#include +#include + +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "common_audio/wav_file.h" +#include "modules/audio_processing/vad/voice_activity_detector.h" +#include "rtc_base/logging.h" + +ABSL_FLAG(std::string, i, "", "Input wav file"); +ABSL_FLAG(std::string, o_probs, "", "VAD probabilities output file"); +ABSL_FLAG(std::string, o_rms, "", "VAD output file"); + +namespace webrtc { +namespace test { +namespace { + +constexpr uint8_t kAudioFrameLengthMilliseconds = 10; +constexpr int kMaxSampleRate = 48000; +constexpr size_t kMaxFrameLen = + kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000; + +int main(int argc, char* argv[]) { + absl::ParseCommandLine(argc, argv); + const std::string input_file = absl::GetFlag(FLAGS_i); + const std::string output_probs_file = absl::GetFlag(FLAGS_o_probs); + const std::string output_file = absl::GetFlag(FLAGS_o_rms); + // Open wav input file and check properties. + WavReader wav_reader(input_file); + if (wav_reader.num_channels() != 1) { + RTC_LOG(LS_ERROR) << "Only mono wav files supported"; + return 1; + } + if (wav_reader.sample_rate() > kMaxSampleRate) { + RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate + << ")"; + return 1; + } + const size_t audio_frame_len = rtc::CheckedDivExact( + kAudioFrameLengthMilliseconds * wav_reader.sample_rate(), 1000); + if (audio_frame_len > kMaxFrameLen) { + RTC_LOG(LS_ERROR) << "The frame size and/or the sample rate are too large."; + return 1; + } + + // Create output file and write header. + std::ofstream out_probs_file(output_probs_file, std::ofstream::binary); + std::ofstream out_rms_file(output_file, std::ofstream::binary); + + // Run VAD and write decisions. + VoiceActivityDetector vad; + std::array samples; + + while (true) { + // Process frame. + const auto read_samples = + wav_reader.ReadSamples(audio_frame_len, samples.data()); + if (read_samples < audio_frame_len) { + break; + } + vad.ProcessChunk(samples.data(), audio_frame_len, wav_reader.sample_rate()); + // Write output. + auto probs = vad.chunkwise_voice_probabilities(); + auto rms = vad.chunkwise_rms(); + RTC_CHECK_EQ(probs.size(), rms.size()); + RTC_CHECK_EQ(sizeof(double), 8); + + for (const auto& p : probs) { + out_probs_file.write(reinterpret_cast(&p), 8); + } + for (const auto& r : rms) { + out_rms_file.write(reinterpret_cast(&r), 8); + } + } + + out_probs_file.close(); + out_rms_file.close(); + return 0; +} + +} // namespace +} // namespace test +} // namespace webrtc + +int main(int argc, char* argv[]) { + return webrtc::test::main(argc, argv); +} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py new file mode 100644 index 0000000000..04aeaa95b9 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py @@ -0,0 +1,100 @@ +# Copyright (c) 2017 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. +"""Class implementing a wrapper for APM simulators. +""" + +import cProfile +import logging +import os +import subprocess + +from . import data_access +from . import exceptions + + +class AudioProcWrapper(object): + """Wrapper for APM simulators. + """ + + DEFAULT_APM_SIMULATOR_BIN_PATH = os.path.abspath( + os.path.join(os.pardir, 'audioproc_f')) + OUTPUT_FILENAME = 'output.wav' + + def __init__(self, simulator_bin_path): + """Ctor. + + Args: + simulator_bin_path: path to the APM simulator binary. + """ + self._simulator_bin_path = simulator_bin_path + self._config = None + self._output_signal_filepath = None + + # Profiler instance to measure running time. + self._profiler = cProfile.Profile() + + @property + def output_filepath(self): + return self._output_signal_filepath + + def Run(self, + config_filepath, + capture_input_filepath, + output_path, + render_input_filepath=None): + """Runs APM simulator. + + Args: + config_filepath: path to the configuration file specifying the arguments + for the APM simulator. + capture_input_filepath: path to the capture audio track input file (aka + forward or near-end). + output_path: path of the audio track output file. + render_input_filepath: path to the render audio track input file (aka + reverse or far-end). + """ + # Init. + self._output_signal_filepath = os.path.join(output_path, + self.OUTPUT_FILENAME) + profiling_stats_filepath = os.path.join(output_path, 'profiling.stats') + + # Skip if the output has already been generated. + if os.path.exists(self._output_signal_filepath) and os.path.exists( + profiling_stats_filepath): + return + + # Load configuration. + self._config = data_access.AudioProcConfigFile.Load(config_filepath) + + # Set remaining parameters. + if not os.path.exists(capture_input_filepath): + raise exceptions.FileNotFoundError( + 'cannot find capture input file') + self._config['-i'] = capture_input_filepath + self._config['-o'] = self._output_signal_filepath + if render_input_filepath is not None: + if not os.path.exists(render_input_filepath): + raise exceptions.FileNotFoundError( + 'cannot find render input file') + self._config['-ri'] = render_input_filepath + + # Build arguments list. + args = [self._simulator_bin_path] + for param_name in self._config: + args.append(param_name) + if self._config[param_name] is not None: + args.append(str(self._config[param_name])) + logging.debug(' '.join(args)) + + # Run. + self._profiler.enable() + subprocess.call(args) + self._profiler.disable() + + # Save profiling stats. + self._profiler.dump_stats(profiling_stats_filepath) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py new file mode 100644 index 0000000000..38aac0cbe2 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py @@ -0,0 +1,243 @@ +# Copyright (c) 2017 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. +"""Imports a filtered subset of the scores and configurations computed +by apm_quality_assessment.py into a pandas data frame. +""" + +import argparse +import glob +import logging +import os +import re +import sys + +try: + import pandas as pd +except ImportError: + logging.critical('Cannot import the third-party Python package pandas') + sys.exit(1) + +from . import data_access as data_access +from . import simulation as sim + +# Compiled regular expressions used to extract score descriptors. +RE_CONFIG_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixApmConfig() + + r'(.+)') +RE_CAPTURE_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixCapture() + + r'(.+)') +RE_RENDER_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixRender() + r'(.+)') +RE_ECHO_SIM_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixEchoSimulator() + + r'(.+)') +RE_TEST_DATA_GEN_NAME = re.compile( + sim.ApmModuleSimulator.GetPrefixTestDataGenerator() + r'(.+)') +RE_TEST_DATA_GEN_PARAMS = re.compile( + sim.ApmModuleSimulator.GetPrefixTestDataGeneratorParameters() + r'(.+)') +RE_SCORE_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixScore() + + r'(.+)(\..+)') + + +def InstanceArgumentsParser(): + """Arguments parser factory. + """ + parser = argparse.ArgumentParser( + description=('Override this description in a user script by changing' + ' `parser.description` of the returned parser.')) + + parser.add_argument('-o', + '--output_dir', + required=True, + help=('the same base path used with the ' + 'apm_quality_assessment tool')) + + parser.add_argument( + '-c', + '--config_names', + type=re.compile, + help=('regular expression to filter the APM configuration' + ' names')) + + parser.add_argument( + '-i', + '--capture_names', + type=re.compile, + help=('regular expression to filter the capture signal ' + 'names')) + + parser.add_argument('-r', + '--render_names', + type=re.compile, + help=('regular expression to filter the render signal ' + 'names')) + + parser.add_argument( + '-e', + '--echo_simulator_names', + type=re.compile, + help=('regular expression to filter the echo simulator ' + 'names')) + + parser.add_argument('-t', + '--test_data_generators', + type=re.compile, + help=('regular expression to filter the test data ' + 'generator names')) + + parser.add_argument( + '-s', + '--eval_scores', + type=re.compile, + help=('regular expression to filter the evaluation score ' + 'names')) + + return parser + + +def _GetScoreDescriptors(score_filepath): + """Extracts a score descriptor from the given score file path. + + Args: + score_filepath: path to the score file. + + Returns: + A tuple of strings (APM configuration name, capture audio track name, + render audio track name, echo simulator name, test data generator name, + test data generator parameters as string, evaluation score name). + """ + fields = score_filepath.split(os.sep)[-7:] + extract_name = lambda index, reg_expr: (reg_expr.match(fields[index]). + groups(0)[0]) + return ( + extract_name(0, RE_CONFIG_NAME), + extract_name(1, RE_CAPTURE_NAME), + extract_name(2, RE_RENDER_NAME), + extract_name(3, RE_ECHO_SIM_NAME), + extract_name(4, RE_TEST_DATA_GEN_NAME), + extract_name(5, RE_TEST_DATA_GEN_PARAMS), + extract_name(6, RE_SCORE_NAME), + ) + + +def _ExcludeScore(config_name, capture_name, render_name, echo_simulator_name, + test_data_gen_name, score_name, args): + """Decides whether excluding a score. + + A set of optional regular expressions in args is used to determine if the + score should be excluded (depending on its |*_name| descriptors). + + Args: + config_name: APM configuration name. + capture_name: capture audio track name. + render_name: render audio track name. + echo_simulator_name: echo simulator name. + test_data_gen_name: test data generator name. + score_name: evaluation score name. + args: parsed arguments. + + Returns: + A boolean. + """ + value_regexpr_pairs = [ + (config_name, args.config_names), + (capture_name, args.capture_names), + (render_name, args.render_names), + (echo_simulator_name, args.echo_simulator_names), + (test_data_gen_name, args.test_data_generators), + (score_name, args.eval_scores), + ] + + # Score accepted if each value matches the corresponding regular expression. + for value, regexpr in value_regexpr_pairs: + if regexpr is None: + continue + if not regexpr.match(value): + return True + + return False + + +def FindScores(src_path, args): + """Given a search path, find scores and return a DataFrame object. + + Args: + src_path: Search path pattern. + args: parsed arguments. + + Returns: + A DataFrame object. + """ + # Get scores. + scores = [] + for score_filepath in glob.iglob(src_path): + # Extract score descriptor fields from the path. + (config_name, capture_name, render_name, echo_simulator_name, + test_data_gen_name, test_data_gen_params, + score_name) = _GetScoreDescriptors(score_filepath) + + # Ignore the score if required. + if _ExcludeScore(config_name, capture_name, render_name, + echo_simulator_name, test_data_gen_name, score_name, + args): + logging.info('ignored score: %s %s %s %s %s %s', config_name, + capture_name, render_name, echo_simulator_name, + test_data_gen_name, score_name) + continue + + # Read metadata and score. + metadata = data_access.Metadata.LoadAudioTestDataPaths( + os.path.split(score_filepath)[0]) + score = data_access.ScoreFile.Load(score_filepath) + + # Add a score with its descriptor fields. + scores.append(( + metadata['clean_capture_input_filepath'], + metadata['echo_free_capture_filepath'], + metadata['echo_filepath'], + metadata['render_filepath'], + metadata['capture_filepath'], + metadata['apm_output_filepath'], + metadata['apm_reference_filepath'], + config_name, + capture_name, + render_name, + echo_simulator_name, + test_data_gen_name, + test_data_gen_params, + score_name, + score, + )) + + return pd.DataFrame(data=scores, + columns=( + 'clean_capture_input_filepath', + 'echo_free_capture_filepath', + 'echo_filepath', + 'render_filepath', + 'capture_filepath', + 'apm_output_filepath', + 'apm_reference_filepath', + 'apm_config', + 'capture', + 'render', + 'echo_simulator', + 'test_data_gen', + 'test_data_gen_params', + 'eval_score_name', + 'score', + )) + + +def ConstructSrcPath(args): + return os.path.join( + args.output_dir, + sim.ApmModuleSimulator.GetPrefixApmConfig() + '*', + sim.ApmModuleSimulator.GetPrefixCapture() + '*', + sim.ApmModuleSimulator.GetPrefixRender() + '*', + sim.ApmModuleSimulator.GetPrefixEchoSimulator() + '*', + sim.ApmModuleSimulator.GetPrefixTestDataGenerator() + '*', + sim.ApmModuleSimulator.GetPrefixTestDataGeneratorParameters() + '*', + sim.ApmModuleSimulator.GetPrefixScore() + '*') diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py new file mode 100644 index 0000000000..c1aebb67f1 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py @@ -0,0 +1,154 @@ +# Copyright (c) 2017 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. +"""Data access utility functions and classes. +""" + +import json +import os + + +def MakeDirectory(path): + """Makes a directory recursively without rising exceptions if existing. + + Args: + path: path to the directory to be created. + """ + if os.path.exists(path): + return + os.makedirs(path) + + +class Metadata(object): + """Data access class to save and load metadata. + """ + + def __init__(self): + pass + + _GENERIC_METADATA_SUFFIX = '.mdata' + _AUDIO_TEST_DATA_FILENAME = 'audio_test_data.json' + + @classmethod + def LoadFileMetadata(cls, filepath): + """Loads generic metadata linked to a file. + + Args: + filepath: path to the metadata file to read. + + Returns: + A dict. + """ + with open(filepath + cls._GENERIC_METADATA_SUFFIX) as f: + return json.load(f) + + @classmethod + def SaveFileMetadata(cls, filepath, metadata): + """Saves generic metadata linked to a file. + + Args: + filepath: path to the metadata file to write. + metadata: a dict. + """ + with open(filepath + cls._GENERIC_METADATA_SUFFIX, 'w') as f: + json.dump(metadata, f) + + @classmethod + def LoadAudioTestDataPaths(cls, metadata_path): + """Loads the input and the reference audio track paths. + + Args: + metadata_path: path to the directory containing the metadata file. + + Returns: + Tuple with the paths to the input and output audio tracks. + """ + metadata_filepath = os.path.join(metadata_path, + cls._AUDIO_TEST_DATA_FILENAME) + with open(metadata_filepath) as f: + return json.load(f) + + @classmethod + def SaveAudioTestDataPaths(cls, output_path, **filepaths): + """Saves the input and the reference audio track paths. + + Args: + output_path: path to the directory containing the metadata file. + + Keyword Args: + filepaths: collection of audio track file paths to save. + """ + output_filepath = os.path.join(output_path, + cls._AUDIO_TEST_DATA_FILENAME) + with open(output_filepath, 'w') as f: + json.dump(filepaths, f) + + +class AudioProcConfigFile(object): + """Data access to load/save APM simulator argument lists. + + The arguments stored in the config files are used to control the APM flags. + """ + + def __init__(self): + pass + + @classmethod + def Load(cls, filepath): + """Loads a configuration file for an APM simulator. + + Args: + filepath: path to the configuration file. + + Returns: + A dict containing the configuration. + """ + with open(filepath) as f: + return json.load(f) + + @classmethod + def Save(cls, filepath, config): + """Saves a configuration file for an APM simulator. + + Args: + filepath: path to the configuration file. + config: a dict containing the configuration. + """ + with open(filepath, 'w') as f: + json.dump(config, f) + + +class ScoreFile(object): + """Data access class to save and load float scalar scores. + """ + + def __init__(self): + pass + + @classmethod + def Load(cls, filepath): + """Loads a score from file. + + Args: + filepath: path to the score file. + + Returns: + A float encoding the score. + """ + with open(filepath) as f: + return float(f.readline().strip()) + + @classmethod + def Save(cls, filepath, score): + """Saves a score into a file. + + Args: + filepath: path to the score file. + score: float encoding the score. + """ + with open(filepath, 'w') as f: + f.write('{0:f}\n'.format(score)) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py new file mode 100644 index 0000000000..65903ea32d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py @@ -0,0 +1,136 @@ +# Copyright (c) 2017 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. +"""Echo path simulation module. +""" + +import hashlib +import os + +from . import signal_processing + + +class EchoPathSimulator(object): + """Abstract class for the echo path simulators. + + In general, an echo path simulator is a function of the render signal and + simulates the propagation of the latter into the microphone (e.g., due to + mechanical or electrical paths). + """ + + NAME = None + REGISTERED_CLASSES = {} + + def __init__(self): + pass + + def Simulate(self, output_path): + """Creates the echo signal and stores it in an audio file (abstract method). + + Args: + output_path: Path in which any output can be saved. + + Returns: + Path to the generated audio track file or None if no echo is present. + """ + raise NotImplementedError() + + @classmethod + def RegisterClass(cls, class_to_register): + """Registers an EchoPathSimulator implementation. + + Decorator to automatically register the classes that extend + EchoPathSimulator. + Example usage: + + @EchoPathSimulator.RegisterClass + class NoEchoPathSimulator(EchoPathSimulator): + pass + """ + cls.REGISTERED_CLASSES[class_to_register.NAME] = class_to_register + return class_to_register + + +@EchoPathSimulator.RegisterClass +class NoEchoPathSimulator(EchoPathSimulator): + """Simulates absence of echo.""" + + NAME = 'noecho' + + def __init__(self): + EchoPathSimulator.__init__(self) + + def Simulate(self, output_path): + return None + + +@EchoPathSimulator.RegisterClass +class LinearEchoPathSimulator(EchoPathSimulator): + """Simulates linear echo path. + + This class applies a given impulse response to the render input and then it + sums the signal to the capture input signal. + """ + + NAME = 'linear' + + def __init__(self, render_input_filepath, impulse_response): + """ + Args: + render_input_filepath: Render audio track file. + impulse_response: list or numpy vector of float values. + """ + EchoPathSimulator.__init__(self) + self._render_input_filepath = render_input_filepath + self._impulse_response = impulse_response + + def Simulate(self, output_path): + """Simulates linear echo path.""" + # Form the file name with a hash of the impulse response. + impulse_response_hash = hashlib.sha256( + str(self._impulse_response).encode('utf-8', 'ignore')).hexdigest() + echo_filepath = os.path.join( + output_path, 'linear_echo_{}.wav'.format(impulse_response_hash)) + + # If the simulated echo audio track file does not exists, create it. + if not os.path.exists(echo_filepath): + render = signal_processing.SignalProcessingUtils.LoadWav( + self._render_input_filepath) + echo = signal_processing.SignalProcessingUtils.ApplyImpulseResponse( + render, self._impulse_response) + signal_processing.SignalProcessingUtils.SaveWav( + echo_filepath, echo) + + return echo_filepath + + +@EchoPathSimulator.RegisterClass +class RecordedEchoPathSimulator(EchoPathSimulator): + """Uses recorded echo. + + This class uses the clean capture input file name to build the file name of + the corresponding recording containing echo (a predefined suffix is used). + Such a file is expected to be already existing. + """ + + NAME = 'recorded' + + _FILE_NAME_SUFFIX = '_echo' + + def __init__(self, render_input_filepath): + EchoPathSimulator.__init__(self) + self._render_input_filepath = render_input_filepath + + def Simulate(self, output_path): + """Uses recorded echo path.""" + path, file_name_ext = os.path.split(self._render_input_filepath) + file_name, file_ext = os.path.splitext(file_name_ext) + echo_filepath = os.path.join( + path, '{}{}{}'.format(file_name, self._FILE_NAME_SUFFIX, file_ext)) + assert os.path.exists(echo_filepath), ( + 'cannot find the echo audio track file {}'.format(echo_filepath)) + return echo_filepath diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py new file mode 100644 index 0000000000..4b46b36b47 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py @@ -0,0 +1,48 @@ +# Copyright (c) 2017 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. +"""Echo path simulation factory module. +""" + +import numpy as np + +from . import echo_path_simulation + + +class EchoPathSimulatorFactory(object): + + # TODO(alessiob): Replace 20 ms delay (at 48 kHz sample rate) with a more + # realistic impulse response. + _LINEAR_ECHO_IMPULSE_RESPONSE = np.array([0.0] * (20 * 48) + [0.15]) + + def __init__(self): + pass + + @classmethod + def GetInstance(cls, echo_path_simulator_class, render_input_filepath): + """Creates an EchoPathSimulator instance given a class object. + + Args: + echo_path_simulator_class: EchoPathSimulator class object (not an + instance). + render_input_filepath: Path to the render audio track file. + + Returns: + An EchoPathSimulator instance. + """ + assert render_input_filepath is not None or ( + echo_path_simulator_class == + echo_path_simulation.NoEchoPathSimulator) + + if echo_path_simulator_class == echo_path_simulation.NoEchoPathSimulator: + return echo_path_simulation.NoEchoPathSimulator() + elif echo_path_simulator_class == ( + echo_path_simulation.LinearEchoPathSimulator): + return echo_path_simulation.LinearEchoPathSimulator( + render_input_filepath, cls._LINEAR_ECHO_IMPULSE_RESPONSE) + else: + return echo_path_simulator_class(render_input_filepath) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py new file mode 100644 index 0000000000..b6cc8abdde --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py @@ -0,0 +1,82 @@ +# Copyright (c) 2017 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. +"""Unit tests for the echo path simulation module. +""" + +import shutil +import os +import tempfile +import unittest + +import pydub + +from . import echo_path_simulation +from . import echo_path_simulation_factory +from . import signal_processing + + +class TestEchoPathSimulators(unittest.TestCase): + """Unit tests for the eval_scores module. + """ + + def setUp(self): + """Creates temporary data.""" + self._tmp_path = tempfile.mkdtemp() + + # Create and save white noise. + silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000) + white_noise = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + silence) + self._audio_track_num_samples = ( + signal_processing.SignalProcessingUtils.CountSamples(white_noise)) + self._audio_track_filepath = os.path.join(self._tmp_path, + 'white_noise.wav') + signal_processing.SignalProcessingUtils.SaveWav( + self._audio_track_filepath, white_noise) + + # Make a copy the white noise audio track file; it will be used by + # echo_path_simulation.RecordedEchoPathSimulator. + shutil.copy(self._audio_track_filepath, + os.path.join(self._tmp_path, 'white_noise_echo.wav')) + + def tearDown(self): + """Recursively deletes temporary folders.""" + shutil.rmtree(self._tmp_path) + + def testRegisteredClasses(self): + # Check that there is at least one registered echo path simulator. + registered_classes = ( + echo_path_simulation.EchoPathSimulator.REGISTERED_CLASSES) + self.assertIsInstance(registered_classes, dict) + self.assertGreater(len(registered_classes), 0) + + # Instance factory. + factory = echo_path_simulation_factory.EchoPathSimulatorFactory() + + # Try each registered echo path simulator. + for echo_path_simulator_name in registered_classes: + simulator = factory.GetInstance( + echo_path_simulator_class=registered_classes[ + echo_path_simulator_name], + render_input_filepath=self._audio_track_filepath) + + echo_filepath = simulator.Simulate(self._tmp_path) + if echo_filepath is None: + self.assertEqual(echo_path_simulation.NoEchoPathSimulator.NAME, + echo_path_simulator_name) + # No other tests in this case. + continue + + # Check that the echo audio track file exists and its length is greater or + # equal to that of the render audio track. + self.assertTrue(os.path.exists(echo_filepath)) + echo = signal_processing.SignalProcessingUtils.LoadWav( + echo_filepath) + self.assertGreaterEqual( + signal_processing.SignalProcessingUtils.CountSamples(echo), + self._audio_track_num_samples) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py new file mode 100644 index 0000000000..59c5f74be4 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py @@ -0,0 +1,427 @@ +# Copyright (c) 2017 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. +"""Evaluation score abstract class and implementations. +""" + +from __future__ import division +import logging +import os +import re +import subprocess +import sys + +try: + import numpy as np +except ImportError: + logging.critical('Cannot import the third-party Python package numpy') + sys.exit(1) + +from . import data_access +from . import exceptions +from . import signal_processing + + +class EvaluationScore(object): + + NAME = None + REGISTERED_CLASSES = {} + + def __init__(self, score_filename_prefix): + self._score_filename_prefix = score_filename_prefix + self._input_signal_metadata = None + self._reference_signal = None + self._reference_signal_filepath = None + self._tested_signal = None + self._tested_signal_filepath = None + self._output_filepath = None + self._score = None + self._render_signal_filepath = None + + @classmethod + def RegisterClass(cls, class_to_register): + """Registers an EvaluationScore implementation. + + Decorator to automatically register the classes that extend EvaluationScore. + Example usage: + + @EvaluationScore.RegisterClass + class AudioLevelScore(EvaluationScore): + pass + """ + cls.REGISTERED_CLASSES[class_to_register.NAME] = class_to_register + return class_to_register + + @property + def output_filepath(self): + return self._output_filepath + + @property + def score(self): + return self._score + + def SetInputSignalMetadata(self, metadata): + """Sets input signal metadata. + + Args: + metadata: dict instance. + """ + self._input_signal_metadata = metadata + + def SetReferenceSignalFilepath(self, filepath): + """Sets the path to the audio track used as reference signal. + + Args: + filepath: path to the reference audio track. + """ + self._reference_signal_filepath = filepath + + def SetTestedSignalFilepath(self, filepath): + """Sets the path to the audio track used as test signal. + + Args: + filepath: path to the test audio track. + """ + self._tested_signal_filepath = filepath + + def SetRenderSignalFilepath(self, filepath): + """Sets the path to the audio track used as render signal. + + Args: + filepath: path to the test audio track. + """ + self._render_signal_filepath = filepath + + def Run(self, output_path): + """Extracts the score for the set test data pair. + + Args: + output_path: path to the directory where the output is written. + """ + self._output_filepath = os.path.join( + output_path, self._score_filename_prefix + self.NAME + '.txt') + try: + # If the score has already been computed, load. + self._LoadScore() + logging.debug('score found and loaded') + except IOError: + # Compute the score. + logging.debug('score not found, compute') + self._Run(output_path) + + def _Run(self, output_path): + # Abstract method. + raise NotImplementedError() + + def _LoadReferenceSignal(self): + assert self._reference_signal_filepath is not None + self._reference_signal = signal_processing.SignalProcessingUtils.LoadWav( + self._reference_signal_filepath) + + def _LoadTestedSignal(self): + assert self._tested_signal_filepath is not None + self._tested_signal = signal_processing.SignalProcessingUtils.LoadWav( + self._tested_signal_filepath) + + def _LoadScore(self): + return data_access.ScoreFile.Load(self._output_filepath) + + def _SaveScore(self): + return data_access.ScoreFile.Save(self._output_filepath, self._score) + + +@EvaluationScore.RegisterClass +class AudioLevelPeakScore(EvaluationScore): + """Peak audio level score. + + Defined as the difference between the peak audio level of the tested and + the reference signals. + + Unit: dB + Ideal: 0 dB + Worst case: +/-inf dB + """ + + NAME = 'audio_level_peak' + + def __init__(self, score_filename_prefix): + EvaluationScore.__init__(self, score_filename_prefix) + + def _Run(self, output_path): + self._LoadReferenceSignal() + self._LoadTestedSignal() + self._score = self._tested_signal.dBFS - self._reference_signal.dBFS + self._SaveScore() + + +@EvaluationScore.RegisterClass +class MeanAudioLevelScore(EvaluationScore): + """Mean audio level score. + + Defined as the difference between the mean audio level of the tested and + the reference signals. + + Unit: dB + Ideal: 0 dB + Worst case: +/-inf dB + """ + + NAME = 'audio_level_mean' + + def __init__(self, score_filename_prefix): + EvaluationScore.__init__(self, score_filename_prefix) + + def _Run(self, output_path): + self._LoadReferenceSignal() + self._LoadTestedSignal() + + dbfs_diffs_sum = 0.0 + seconds = min(len(self._tested_signal), len( + self._reference_signal)) // 1000 + for t in range(seconds): + t0 = t * seconds + t1 = t0 + seconds + dbfs_diffs_sum += (self._tested_signal[t0:t1].dBFS - + self._reference_signal[t0:t1].dBFS) + self._score = dbfs_diffs_sum / float(seconds) + self._SaveScore() + + +@EvaluationScore.RegisterClass +class EchoMetric(EvaluationScore): + """Echo score. + + Proportion of detected echo. + + Unit: ratio + Ideal: 0 + Worst case: 1 + """ + + NAME = 'echo_metric' + + def __init__(self, score_filename_prefix, echo_detector_bin_filepath): + EvaluationScore.__init__(self, score_filename_prefix) + + # POLQA binary file path. + self._echo_detector_bin_filepath = echo_detector_bin_filepath + if not os.path.exists(self._echo_detector_bin_filepath): + logging.error('cannot find EchoMetric tool binary file') + raise exceptions.FileNotFoundError() + + self._echo_detector_bin_path, _ = os.path.split( + self._echo_detector_bin_filepath) + + def _Run(self, output_path): + echo_detector_out_filepath = os.path.join(output_path, + 'echo_detector.out') + if os.path.exists(echo_detector_out_filepath): + os.unlink(echo_detector_out_filepath) + + logging.debug("Render signal filepath: %s", + self._render_signal_filepath) + if not os.path.exists(self._render_signal_filepath): + logging.error( + "Render input required for evaluating the echo metric.") + + args = [ + self._echo_detector_bin_filepath, '--output_file', + echo_detector_out_filepath, '--', '-i', + self._tested_signal_filepath, '-ri', self._render_signal_filepath + ] + logging.debug(' '.join(args)) + subprocess.call(args, cwd=self._echo_detector_bin_path) + + # Parse Echo detector tool output and extract the score. + self._score = self._ParseOutputFile(echo_detector_out_filepath) + self._SaveScore() + + @classmethod + def _ParseOutputFile(cls, echo_metric_file_path): + """ + Parses the POLQA tool output formatted as a table ('-t' option). + + Args: + polqa_out_filepath: path to the POLQA tool output file. + + Returns: + The score as a number in [0, 1]. + """ + with open(echo_metric_file_path) as f: + return float(f.read()) + + +@EvaluationScore.RegisterClass +class PolqaScore(EvaluationScore): + """POLQA score. + + See http://www.polqa.info/. + + Unit: MOS + Ideal: 4.5 + Worst case: 1.0 + """ + + NAME = 'polqa' + + def __init__(self, score_filename_prefix, polqa_bin_filepath): + EvaluationScore.__init__(self, score_filename_prefix) + + # POLQA binary file path. + self._polqa_bin_filepath = polqa_bin_filepath + if not os.path.exists(self._polqa_bin_filepath): + logging.error('cannot find POLQA tool binary file') + raise exceptions.FileNotFoundError() + + # Path to the POLQA directory with binary and license files. + self._polqa_tool_path, _ = os.path.split(self._polqa_bin_filepath) + + def _Run(self, output_path): + polqa_out_filepath = os.path.join(output_path, 'polqa.out') + if os.path.exists(polqa_out_filepath): + os.unlink(polqa_out_filepath) + + args = [ + self._polqa_bin_filepath, + '-t', + '-q', + '-Overwrite', + '-Ref', + self._reference_signal_filepath, + '-Test', + self._tested_signal_filepath, + '-LC', + 'NB', + '-Out', + polqa_out_filepath, + ] + logging.debug(' '.join(args)) + subprocess.call(args, cwd=self._polqa_tool_path) + + # Parse POLQA tool output and extract the score. + polqa_output = self._ParseOutputFile(polqa_out_filepath) + self._score = float(polqa_output['PolqaScore']) + + self._SaveScore() + + @classmethod + def _ParseOutputFile(cls, polqa_out_filepath): + """ + Parses the POLQA tool output formatted as a table ('-t' option). + + Args: + polqa_out_filepath: path to the POLQA tool output file. + + Returns: + A dict. + """ + data = [] + with open(polqa_out_filepath) as f: + for line in f: + line = line.strip() + if len(line) == 0 or line.startswith('*'): + # Ignore comments. + continue + # Read fields. + data.append(re.split(r'\t+', line)) + + # Two rows expected (header and values). + assert len(data) == 2, 'Cannot parse POLQA output' + number_of_fields = len(data[0]) + assert number_of_fields == len(data[1]) + + # Build and return a dictionary with field names (header) as keys and the + # corresponding field values as values. + return { + data[0][index]: data[1][index] + for index in range(number_of_fields) + } + + +@EvaluationScore.RegisterClass +class TotalHarmonicDistorsionScore(EvaluationScore): + """Total harmonic distorsion plus noise score. + + Total harmonic distorsion plus noise score. + See "https://en.wikipedia.org/wiki/Total_harmonic_distortion#THD.2BN". + + Unit: -. + Ideal: 0. + Worst case: +inf + """ + + NAME = 'thd' + + def __init__(self, score_filename_prefix): + EvaluationScore.__init__(self, score_filename_prefix) + self._input_frequency = None + + def _Run(self, output_path): + self._CheckInputSignal() + + self._LoadTestedSignal() + if self._tested_signal.channels != 1: + raise exceptions.EvaluationScoreException( + 'unsupported number of channels') + samples = signal_processing.SignalProcessingUtils.AudioSegmentToRawData( + self._tested_signal) + + # Init. + num_samples = len(samples) + duration = len(self._tested_signal) / 1000.0 + scaling = 2.0 / num_samples + max_freq = self._tested_signal.frame_rate / 2 + f0_freq = float(self._input_frequency) + t = np.linspace(0, duration, num_samples) + + # Analyze harmonics. + b_terms = [] + n = 1 + while f0_freq * n < max_freq: + x_n = np.sum( + samples * np.sin(2.0 * np.pi * n * f0_freq * t)) * scaling + y_n = np.sum( + samples * np.cos(2.0 * np.pi * n * f0_freq * t)) * scaling + b_terms.append(np.sqrt(x_n**2 + y_n**2)) + n += 1 + + output_without_fundamental = samples - b_terms[0] * np.sin( + 2.0 * np.pi * f0_freq * t) + distortion_and_noise = np.sqrt( + np.sum(output_without_fundamental**2) * np.pi * scaling) + + # TODO(alessiob): Fix or remove if not needed. + # thd = np.sqrt(np.sum(b_terms[1:]**2)) / b_terms[0] + + # TODO(alessiob): Check the range of `thd_plus_noise` and update the class + # docstring above if accordingly. + thd_plus_noise = distortion_and_noise / b_terms[0] + + self._score = thd_plus_noise + self._SaveScore() + + def _CheckInputSignal(self): + # Check input signal and get properties. + try: + if self._input_signal_metadata['signal'] != 'pure_tone': + raise exceptions.EvaluationScoreException( + 'The THD score requires a pure tone as input signal') + self._input_frequency = self._input_signal_metadata['frequency'] + if self._input_signal_metadata[ + 'test_data_gen_name'] != 'identity' or ( + self._input_signal_metadata['test_data_gen_config'] != + 'default'): + raise exceptions.EvaluationScoreException( + 'The THD score cannot be used with any test data generator other ' + 'than "identity"') + except TypeError: + raise exceptions.EvaluationScoreException( + 'The THD score requires an input signal with associated metadata' + ) + except KeyError: + raise exceptions.EvaluationScoreException( + 'Invalid input signal metadata to compute the THD score') diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py new file mode 100644 index 0000000000..5749a8924b --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py @@ -0,0 +1,55 @@ +# Copyright (c) 2017 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. +"""EvaluationScore factory class. +""" + +import logging + +from . import exceptions +from . import eval_scores + + +class EvaluationScoreWorkerFactory(object): + """Factory class used to instantiate evaluation score workers. + + The ctor gets the parametrs that are used to instatiate the evaluation score + workers. + """ + + def __init__(self, polqa_tool_bin_path, echo_metric_tool_bin_path): + self._score_filename_prefix = None + self._polqa_tool_bin_path = polqa_tool_bin_path + self._echo_metric_tool_bin_path = echo_metric_tool_bin_path + + def SetScoreFilenamePrefix(self, prefix): + self._score_filename_prefix = prefix + + def GetInstance(self, evaluation_score_class): + """Creates an EvaluationScore instance given a class object. + + Args: + evaluation_score_class: EvaluationScore class object (not an instance). + + Returns: + An EvaluationScore instance. + """ + if self._score_filename_prefix is None: + raise exceptions.InitializationException( + 'The score file name prefix for evaluation score workers is not set' + ) + logging.debug('factory producing a %s evaluation score', + evaluation_score_class) + + if evaluation_score_class == eval_scores.PolqaScore: + return eval_scores.PolqaScore(self._score_filename_prefix, + self._polqa_tool_bin_path) + elif evaluation_score_class == eval_scores.EchoMetric: + return eval_scores.EchoMetric(self._score_filename_prefix, + self._echo_metric_tool_bin_path) + else: + return evaluation_score_class(self._score_filename_prefix) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py new file mode 100644 index 0000000000..12e043320e --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py @@ -0,0 +1,137 @@ +# Copyright (c) 2017 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. +"""Unit tests for the eval_scores module. +""" + +import os +import shutil +import tempfile +import unittest + +import pydub + +from . import data_access +from . import eval_scores +from . import eval_scores_factory +from . import signal_processing + + +class TestEvalScores(unittest.TestCase): + """Unit tests for the eval_scores module. + """ + + def setUp(self): + """Create temporary output folder and two audio track files.""" + self._output_path = tempfile.mkdtemp() + + # Create fake reference and tested (i.e., APM output) audio track files. + silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000) + fake_reference_signal = (signal_processing.SignalProcessingUtils. + GenerateWhiteNoise(silence)) + fake_tested_signal = (signal_processing.SignalProcessingUtils. + GenerateWhiteNoise(silence)) + + # Save fake audio tracks. + self._fake_reference_signal_filepath = os.path.join( + self._output_path, 'fake_ref.wav') + signal_processing.SignalProcessingUtils.SaveWav( + self._fake_reference_signal_filepath, fake_reference_signal) + self._fake_tested_signal_filepath = os.path.join( + self._output_path, 'fake_test.wav') + signal_processing.SignalProcessingUtils.SaveWav( + self._fake_tested_signal_filepath, fake_tested_signal) + + def tearDown(self): + """Recursively delete temporary folder.""" + shutil.rmtree(self._output_path) + + def testRegisteredClasses(self): + # Evaluation score names to exclude (tested separately). + exceptions = ['thd', 'echo_metric'] + + # Preliminary check. + self.assertTrue(os.path.exists(self._output_path)) + + # Check that there is at least one registered evaluation score worker. + registered_classes = eval_scores.EvaluationScore.REGISTERED_CLASSES + self.assertIsInstance(registered_classes, dict) + self.assertGreater(len(registered_classes), 0) + + # Instance evaluation score workers factory with fake dependencies. + eval_score_workers_factory = ( + eval_scores_factory.EvaluationScoreWorkerFactory( + polqa_tool_bin_path=os.path.join( + os.path.dirname(os.path.abspath(__file__)), 'fake_polqa'), + echo_metric_tool_bin_path=None)) + eval_score_workers_factory.SetScoreFilenamePrefix('scores-') + + # Try each registered evaluation score worker. + for eval_score_name in registered_classes: + if eval_score_name in exceptions: + continue + + # Instance evaluation score worker. + eval_score_worker = eval_score_workers_factory.GetInstance( + registered_classes[eval_score_name]) + + # Set fake input metadata and reference and test file paths, then run. + eval_score_worker.SetReferenceSignalFilepath( + self._fake_reference_signal_filepath) + eval_score_worker.SetTestedSignalFilepath( + self._fake_tested_signal_filepath) + eval_score_worker.Run(self._output_path) + + # Check output. + score = data_access.ScoreFile.Load( + eval_score_worker.output_filepath) + self.assertTrue(isinstance(score, float)) + + def testTotalHarmonicDistorsionScore(self): + # Init. + pure_tone_freq = 5000.0 + eval_score_worker = eval_scores.TotalHarmonicDistorsionScore('scores-') + eval_score_worker.SetInputSignalMetadata({ + 'signal': + 'pure_tone', + 'frequency': + pure_tone_freq, + 'test_data_gen_name': + 'identity', + 'test_data_gen_config': + 'default', + }) + template = pydub.AudioSegment.silent(duration=1000, frame_rate=48000) + + # Create 3 test signals: pure tone, pure tone + white noise, white noise + # only. + pure_tone = signal_processing.SignalProcessingUtils.GeneratePureTone( + template, pure_tone_freq) + white_noise = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + template) + noisy_tone = signal_processing.SignalProcessingUtils.MixSignals( + pure_tone, white_noise) + + # Compute scores for increasingly distorted pure tone signals. + scores = [None, None, None] + for index, tested_signal in enumerate( + [pure_tone, noisy_tone, white_noise]): + # Save signal. + tmp_filepath = os.path.join(self._output_path, 'tmp_thd.wav') + signal_processing.SignalProcessingUtils.SaveWav( + tmp_filepath, tested_signal) + + # Compute score. + eval_score_worker.SetTestedSignalFilepath(tmp_filepath) + eval_score_worker.Run(self._output_path) + scores[index] = eval_score_worker.score + + # Remove output file to avoid caching. + os.remove(eval_score_worker.output_filepath) + + # Validate scores (lowest score with a pure tone). + self.assertTrue(all([scores[i + 1] > scores[i] for i in range(2)])) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py new file mode 100644 index 0000000000..2599085329 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py @@ -0,0 +1,57 @@ +# Copyright (c) 2017 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. +"""Evaluator of the APM module. +""" + +import logging + + +class ApmModuleEvaluator(object): + """APM evaluator class. + """ + + def __init__(self): + pass + + @classmethod + def Run(cls, evaluation_score_workers, apm_input_metadata, + apm_output_filepath, reference_input_filepath, + render_input_filepath, output_path): + """Runs the evaluation. + + Iterates over the given evaluation score workers. + + Args: + evaluation_score_workers: list of EvaluationScore instances. + apm_input_metadata: dictionary with metadata of the APM input. + apm_output_filepath: path to the audio track file with the APM output. + reference_input_filepath: path to the reference audio track file. + output_path: output path. + + Returns: + A dict of evaluation score name and score pairs. + """ + # Init. + scores = {} + + for evaluation_score_worker in evaluation_score_workers: + logging.info(' computing <%s> score', + evaluation_score_worker.NAME) + evaluation_score_worker.SetInputSignalMetadata(apm_input_metadata) + evaluation_score_worker.SetReferenceSignalFilepath( + reference_input_filepath) + evaluation_score_worker.SetTestedSignalFilepath( + apm_output_filepath) + evaluation_score_worker.SetRenderSignalFilepath( + render_input_filepath) + + evaluation_score_worker.Run(output_path) + scores[ + evaluation_score_worker.NAME] = evaluation_score_worker.score + + return scores diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py new file mode 100644 index 0000000000..893901d359 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py @@ -0,0 +1,45 @@ +# Copyright (c) 2017 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. +"""Exception classes. +""" + + +class FileNotFoundError(Exception): + """File not found exception. + """ + pass + + +class SignalProcessingException(Exception): + """Signal processing exception. + """ + pass + + +class InputMixerException(Exception): + """Input mixer exception. + """ + pass + + +class InputSignalCreatorException(Exception): + """Input signal creator exception. + """ + pass + + +class EvaluationScoreException(Exception): + """Evaluation score exception. + """ + pass + + +class InitializationException(Exception): + """Initialization exception. + """ + pass diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py new file mode 100644 index 0000000000..fe3a6c7cb9 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py @@ -0,0 +1,426 @@ +# Copyright (c) 2017 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. + +import functools +import hashlib +import logging +import os +import re +import sys + +try: + import csscompressor +except ImportError: + logging.critical( + 'Cannot import the third-party Python package csscompressor') + sys.exit(1) + +try: + import jsmin +except ImportError: + logging.critical('Cannot import the third-party Python package jsmin') + sys.exit(1) + + +class HtmlExport(object): + """HTML exporter class for APM quality scores.""" + + _NEW_LINE = '\n' + + # CSS and JS file paths. + _PATH = os.path.dirname(os.path.realpath(__file__)) + _CSS_FILEPATH = os.path.join(_PATH, 'results.css') + _CSS_MINIFIED = True + _JS_FILEPATH = os.path.join(_PATH, 'results.js') + _JS_MINIFIED = True + + def __init__(self, output_filepath): + self._scores_data_frame = None + self._output_filepath = output_filepath + + def Export(self, scores_data_frame): + """Exports scores into an HTML file. + + Args: + scores_data_frame: DataFrame instance. + """ + self._scores_data_frame = scores_data_frame + html = [ + '', + self._BuildHeader(), + (''), '', + self._BuildBody(), '', '' + ] + self._Save(self._output_filepath, self._NEW_LINE.join(html)) + + def _BuildHeader(self): + """Builds the section of the HTML file. + + The header contains the page title and either embedded or linked CSS and JS + files. + + Returns: + A string with ... HTML. + """ + html = ['', 'Results'] + + # Add Material Design hosted libs. + html.append('') + html.append( + '') + html.append( + '') + html.append('') + + # Embed custom JavaScript and CSS files. + html.append('') + html.append('') + + html.append('') + + return self._NEW_LINE.join(html) + + def _BuildBody(self): + """Builds the content of the section.""" + score_names = self._scores_data_frame[ + 'eval_score_name'].drop_duplicates().values.tolist() + + html = [ + ('
'), + '
', + '
', + 'APM QA results ({})'.format( + self._output_filepath), + '
', + ] + + # Tab selectors. + html.append('
') + for tab_index, score_name in enumerate(score_names): + is_active = tab_index == 0 + html.append('' + '{}'.format(tab_index, + ' is-active' if is_active else '', + self._FormatName(score_name))) + html.append('
') + + html.append('
') + html.append( + '
') + + # Tabs content. + for tab_index, score_name in enumerate(score_names): + html.append('
'.format( + ' is-active' if is_active else '', tab_index)) + html.append('
') + html.append( + self._BuildScoreTab(score_name, ('s{}'.format(tab_index), ))) + html.append('
') + html.append('
') + + html.append('
') + html.append('
') + + # Add snackbar for notifications. + html.append( + '
' + '
' + '' + '
') + + return self._NEW_LINE.join(html) + + def _BuildScoreTab(self, score_name, anchor_data): + """Builds the content of a tab.""" + # Find unique values. + scores = self._scores_data_frame[ + self._scores_data_frame.eval_score_name == score_name] + apm_configs = sorted(self._FindUniqueTuples(scores, ['apm_config'])) + test_data_gen_configs = sorted( + self._FindUniqueTuples(scores, + ['test_data_gen', 'test_data_gen_params'])) + + html = [ + '
', + '
', + '
', + (''), + ] + + # Header. + html.append('') + for test_data_gen_info in test_data_gen_configs: + html.append(''.format( + self._FormatName(test_data_gen_info[0]), + test_data_gen_info[1])) + html.append('') + + # Body. + html.append('') + for apm_config in apm_configs: + html.append('') + for test_data_gen_info in test_data_gen_configs: + dialog_id = self._ScoreStatsInspectorDialogId( + score_name, apm_config[0], test_data_gen_info[0], + test_data_gen_info[1]) + html.append( + ''. + format( + dialog_id, + self._BuildScoreTableCell(score_name, + test_data_gen_info[0], + test_data_gen_info[1], + apm_config[0]))) + html.append('') + html.append('') + + html.append( + '
APM config / Test data generator{} {}
' + self._FormatName(apm_config[0]) + '{}
') + + html.append( + self._BuildScoreStatsInspectorDialogs(score_name, apm_configs, + test_data_gen_configs, + anchor_data)) + + return self._NEW_LINE.join(html) + + def _BuildScoreTableCell(self, score_name, test_data_gen, + test_data_gen_params, apm_config): + """Builds the content of a table cell for a score table.""" + scores = self._SliceDataForScoreTableCell(score_name, apm_config, + test_data_gen, + test_data_gen_params) + stats = self._ComputeScoreStats(scores) + + html = [] + items_id_prefix = (score_name + test_data_gen + test_data_gen_params + + apm_config) + if stats['count'] == 1: + # Show the only available score. + item_id = hashlib.md5(items_id_prefix.encode('utf-8')).hexdigest() + html.append('
{1:f}
'.format( + item_id, scores['score'].mean())) + html.append( + '
{}' + '
'.format(item_id, 'single value')) + else: + # Show stats. + for stat_name in ['min', 'max', 'mean', 'std dev']: + item_id = hashlib.md5( + (items_id_prefix + stat_name).encode('utf-8')).hexdigest() + html.append('
{1:f}
'.format( + item_id, stats[stat_name])) + html.append( + '
{}' + '
'.format(item_id, stat_name)) + + return self._NEW_LINE.join(html) + + def _BuildScoreStatsInspectorDialogs(self, score_name, apm_configs, + test_data_gen_configs, anchor_data): + """Builds a set of score stats inspector dialogs.""" + html = [] + for apm_config in apm_configs: + for test_data_gen_info in test_data_gen_configs: + dialog_id = self._ScoreStatsInspectorDialogId( + score_name, apm_config[0], test_data_gen_info[0], + test_data_gen_info[1]) + + html.append(''.format(dialog_id)) + + # Content. + html.append('
') + html.append( + '
APM config preset: {}
' + 'Test data generator: {} ({})
'. + format(self._FormatName(apm_config[0]), + self._FormatName(test_data_gen_info[0]), + test_data_gen_info[1])) + html.append( + self._BuildScoreStatsInspectorDialog( + score_name, apm_config[0], test_data_gen_info[0], + test_data_gen_info[1], anchor_data + (dialog_id, ))) + html.append('
') + + # Actions. + html.append('
') + html.append('') + html.append('
') + + html.append('') + + return self._NEW_LINE.join(html) + + def _BuildScoreStatsInspectorDialog(self, score_name, apm_config, + test_data_gen, test_data_gen_params, + anchor_data): + """Builds one score stats inspector dialog.""" + scores = self._SliceDataForScoreTableCell(score_name, apm_config, + test_data_gen, + test_data_gen_params) + + capture_render_pairs = sorted( + self._FindUniqueTuples(scores, ['capture', 'render'])) + echo_simulators = sorted( + self._FindUniqueTuples(scores, ['echo_simulator'])) + + html = [ + '' + ] + + # Header. + html.append('') + for echo_simulator in echo_simulators: + html.append('') + html.append('') + + # Body. + html.append('') + for row, (capture, render) in enumerate(capture_render_pairs): + html.append(''.format( + capture, render)) + for col, echo_simulator in enumerate(echo_simulators): + score_tuple = self._SliceDataForScoreStatsTableCell( + scores, capture, render, echo_simulator[0]) + cell_class = 'r{}c{}'.format(row, col) + html.append(''.format( + cell_class, + self._BuildScoreStatsInspectorTableCell( + score_tuple, anchor_data + (cell_class, )))) + html.append('') + html.append('') + + html.append('
Capture-Render / Echo simulator' + self._FormatName(echo_simulator[0]) + '
{}
{}
{}
') + + # Placeholder for the audio inspector. + html.append('
') + + return self._NEW_LINE.join(html) + + def _BuildScoreStatsInspectorTableCell(self, score_tuple, anchor_data): + """Builds the content of a cell of a score stats inspector.""" + anchor = '&'.join(anchor_data) + html = [('
{}
' + '').format(score_tuple.score, anchor)] + + # Add all the available file paths as hidden data. + for field_name in score_tuple.keys(): + if field_name.endswith('_filepath'): + html.append( + ''.format( + field_name, score_tuple[field_name])) + + return self._NEW_LINE.join(html) + + def _SliceDataForScoreTableCell(self, score_name, apm_config, + test_data_gen, test_data_gen_params): + """Slices `self._scores_data_frame` to extract the data for a tab.""" + masks = [] + masks.append(self._scores_data_frame.eval_score_name == score_name) + masks.append(self._scores_data_frame.apm_config == apm_config) + masks.append(self._scores_data_frame.test_data_gen == test_data_gen) + masks.append(self._scores_data_frame.test_data_gen_params == + test_data_gen_params) + mask = functools.reduce((lambda i1, i2: i1 & i2), masks) + del masks + return self._scores_data_frame[mask] + + @classmethod + def _SliceDataForScoreStatsTableCell(cls, scores, capture, render, + echo_simulator): + """Slices `scores` to extract the data for a tab.""" + masks = [] + + masks.append(scores.capture == capture) + masks.append(scores.render == render) + masks.append(scores.echo_simulator == echo_simulator) + mask = functools.reduce((lambda i1, i2: i1 & i2), masks) + del masks + + sliced_data = scores[mask] + assert len(sliced_data) == 1, 'single score is expected' + return sliced_data.iloc[0] + + @classmethod + def _FindUniqueTuples(cls, data_frame, fields): + """Slices `data_frame` to a list of fields and finds unique tuples.""" + return data_frame[fields].drop_duplicates().values.tolist() + + @classmethod + def _ComputeScoreStats(cls, data_frame): + """Computes score stats.""" + scores = data_frame['score'] + return { + 'count': scores.count(), + 'min': scores.min(), + 'max': scores.max(), + 'mean': scores.mean(), + 'std dev': scores.std(), + } + + @classmethod + def _ScoreStatsInspectorDialogId(cls, score_name, apm_config, + test_data_gen, test_data_gen_params): + """Assigns a unique name to a dialog.""" + return 'score-stats-dialog-' + hashlib.md5( + 'score-stats-inspector-{}-{}-{}-{}'.format( + score_name, apm_config, test_data_gen, + test_data_gen_params).encode('utf-8')).hexdigest() + + @classmethod + def _Save(cls, output_filepath, html): + """Writes the HTML file. + + Args: + output_filepath: output file path. + html: string with the HTML content. + """ + with open(output_filepath, 'w') as f: + f.write(html) + + @classmethod + def _FormatName(cls, name): + """Formats a name. + + Args: + name: a string. + + Returns: + A copy of name in which underscores and dashes are replaced with a space. + """ + return re.sub(r'[_\-]', ' ', name) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py new file mode 100644 index 0000000000..412aa7c4e7 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py @@ -0,0 +1,86 @@ +# Copyright (c) 2017 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. +"""Unit tests for the export module. +""" + +import logging +import os +import shutil +import tempfile +import unittest + +import pyquery as pq + +from . import audioproc_wrapper +from . import collect_data +from . import eval_scores_factory +from . import evaluation +from . import export +from . import simulation +from . import test_data_generation_factory + + +class TestExport(unittest.TestCase): + """Unit tests for the export module. + """ + + _CLEAN_TMP_OUTPUT = True + + def setUp(self): + """Creates temporary data to export.""" + self._tmp_path = tempfile.mkdtemp() + + # Run a fake experiment to produce data to export. + simulator = simulation.ApmModuleSimulator( + test_data_generator_factory=( + test_data_generation_factory.TestDataGeneratorFactory( + aechen_ir_database_path='', + noise_tracks_path='', + copy_with_identity=False)), + evaluation_score_factory=( + eval_scores_factory.EvaluationScoreWorkerFactory( + polqa_tool_bin_path=os.path.join( + os.path.dirname(os.path.abspath(__file__)), + 'fake_polqa'), + echo_metric_tool_bin_path=None)), + ap_wrapper=audioproc_wrapper.AudioProcWrapper( + audioproc_wrapper.AudioProcWrapper. + DEFAULT_APM_SIMULATOR_BIN_PATH), + evaluator=evaluation.ApmModuleEvaluator()) + simulator.Run( + config_filepaths=['apm_configs/default.json'], + capture_input_filepaths=[ + os.path.join(self._tmp_path, 'pure_tone-440_1000.wav'), + os.path.join(self._tmp_path, 'pure_tone-880_1000.wav'), + ], + test_data_generator_names=['identity', 'white_noise'], + eval_score_names=['audio_level_peak', 'audio_level_mean'], + output_dir=self._tmp_path) + + # Export results. + p = collect_data.InstanceArgumentsParser() + args = p.parse_args(['--output_dir', self._tmp_path]) + src_path = collect_data.ConstructSrcPath(args) + self._data_to_export = collect_data.FindScores(src_path, args) + + def tearDown(self): + """Recursively deletes temporary folders.""" + if self._CLEAN_TMP_OUTPUT: + shutil.rmtree(self._tmp_path) + else: + logging.warning(self.id() + ' did not clean the temporary path ' + + (self._tmp_path)) + + def testCreateHtmlReport(self): + fn_out = os.path.join(self._tmp_path, 'results.html') + exporter = export.HtmlExport(fn_out) + exporter.Export(self._data_to_export) + + document = pq.PyQuery(filename=fn_out) + self.assertIsInstance(document, pq.PyQuery) + # TODO(alessiob): Use PyQuery API to check the HTML file. diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py new file mode 100644 index 0000000000..a7db7b4840 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py @@ -0,0 +1,75 @@ +# Copyright (c) 2017 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. + +from __future__ import division + +import logging +import os +import subprocess +import shutil +import sys +import tempfile + +try: + import numpy as np +except ImportError: + logging.critical('Cannot import the third-party Python package numpy') + sys.exit(1) + +from . import signal_processing + + +class ExternalVad(object): + def __init__(self, path_to_binary, name): + """Args: + path_to_binary: path to binary that accepts '-i ', '-o + '. There must be one float value per + 10ms audio + name: a name to identify the external VAD. Used for saving + the output as extvad_output-. + """ + self._path_to_binary = path_to_binary + self.name = name + assert os.path.exists(self._path_to_binary), (self._path_to_binary) + self._vad_output = None + + def Run(self, wav_file_path): + _signal = signal_processing.SignalProcessingUtils.LoadWav( + wav_file_path) + if _signal.channels != 1: + raise NotImplementedError('Multiple-channel' + ' annotations not implemented') + if _signal.frame_rate != 48000: + raise NotImplementedError('Frame rates ' + 'other than 48000 not implemented') + + tmp_path = tempfile.mkdtemp() + try: + output_file_path = os.path.join(tmp_path, self.name + '_vad.tmp') + subprocess.call([ + self._path_to_binary, '-i', wav_file_path, '-o', + output_file_path + ]) + self._vad_output = np.fromfile(output_file_path, np.float32) + except Exception as e: + logging.error('Error while running the ' + self.name + ' VAD (' + + e.message + ')') + finally: + if os.path.exists(tmp_path): + shutil.rmtree(tmp_path) + + def GetVadOutput(self): + assert self._vad_output is not None + return self._vad_output + + @classmethod + def ConstructVadDict(cls, vad_paths, vad_names): + external_vads = {} + for path, name in zip(vad_paths, vad_names): + external_vads[name] = ExternalVad(path, name) + return external_vads diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py new file mode 100755 index 0000000000..f679f8c94a --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py @@ -0,0 +1,25 @@ +#!/usr/bin/python +# Copyright (c) 2017 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. +import argparse +import numpy as np + + +def main(): + parser = argparse.ArgumentParser() + parser.add_argument('-i', required=True) + parser.add_argument('-o', required=True) + + args = parser.parse_args() + + array = np.arange(100, dtype=np.float32) + array.tofile(open(args.o, 'w')) + + +if __name__ == '__main__': + main() diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc new file mode 100644 index 0000000000..6f3b2d1dd7 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc @@ -0,0 +1,56 @@ +/* + * Copyright (c) 2017 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 +#include +#include + +#include "absl/strings/string_view.h" +#include "rtc_base/checks.h" + +namespace webrtc { +namespace test { +namespace { + +const char* const kErrorMessage = "-Out /path/to/output/file is mandatory"; + +// Writes fake output intended to be parsed by +// quality_assessment.eval_scores.PolqaScore. +void WriteOutputFile(absl::string_view output_file_path) { + RTC_CHECK_NE(output_file_path, ""); + std::ofstream out(std::string{output_file_path}); + RTC_CHECK(!out.bad()); + out << "* Fake Polqa output" << std::endl; + out << "FakeField1\tPolqaScore\tFakeField2" << std::endl; + out << "FakeValue1\t3.25\tFakeValue2" << std::endl; + out.close(); +} + +} // namespace + +int main(int argc, char* argv[]) { + // Find "-Out" and use its next argument as output file path. + RTC_CHECK_GE(argc, 3) << kErrorMessage; + const std::string kSoughtFlagName = "-Out"; + for (int i = 1; i < argc - 1; ++i) { + if (kSoughtFlagName.compare(argv[i]) == 0) { + WriteOutputFile(argv[i + 1]); + return 0; + } + } + RTC_FATAL() << kErrorMessage; +} + +} // namespace test +} // namespace webrtc + +int main(int argc, char* argv[]) { + return webrtc::test::main(argc, argv); +} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py new file mode 100644 index 0000000000..af022bd461 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py @@ -0,0 +1,97 @@ +# Copyright (c) 2017 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. +"""Input mixer module. +""" + +import logging +import os + +from . import exceptions +from . import signal_processing + + +class ApmInputMixer(object): + """Class to mix a set of audio segments down to the APM input.""" + + _HARD_CLIPPING_LOG_MSG = 'hard clipping detected in the mixed signal' + + def __init__(self): + pass + + @classmethod + def HardClippingLogMessage(cls): + """Returns the log message used when hard clipping is detected in the mix. + + This method is mainly intended to be used by the unit tests. + """ + return cls._HARD_CLIPPING_LOG_MSG + + @classmethod + def Mix(cls, output_path, capture_input_filepath, echo_filepath): + """Mixes capture and echo. + + Creates the overall capture input for APM by mixing the "echo-free" capture + signal with the echo signal (e.g., echo simulated via the + echo_path_simulation module). + + The echo signal cannot be shorter than the capture signal and the generated + mix will have the same duration of the capture signal. The latter property + is enforced in order to let the input of APM and the reference signal + created by TestDataGenerator have the same length (required for the + evaluation step). + + Hard-clipping may occur in the mix; a warning is raised when this happens. + + If `echo_filepath` is None, nothing is done and `capture_input_filepath` is + returned. + + Args: + speech: AudioSegment instance. + echo_path: AudioSegment instance or None. + + Returns: + Path to the mix audio track file. + """ + if echo_filepath is None: + return capture_input_filepath + + # Build the mix output file name as a function of the echo file name. + # This ensures that if the internal parameters of the echo path simulator + # change, no erroneous cache hit occurs. + echo_file_name, _ = os.path.splitext(os.path.split(echo_filepath)[1]) + capture_input_file_name, _ = os.path.splitext( + os.path.split(capture_input_filepath)[1]) + mix_filepath = os.path.join( + output_path, + 'mix_capture_{}_{}.wav'.format(capture_input_file_name, + echo_file_name)) + + # Create the mix if not done yet. + mix = None + if not os.path.exists(mix_filepath): + echo_free_capture = signal_processing.SignalProcessingUtils.LoadWav( + capture_input_filepath) + echo = signal_processing.SignalProcessingUtils.LoadWav( + echo_filepath) + + if signal_processing.SignalProcessingUtils.CountSamples(echo) < ( + signal_processing.SignalProcessingUtils.CountSamples( + echo_free_capture)): + raise exceptions.InputMixerException( + 'echo cannot be shorter than capture') + + mix = echo_free_capture.overlay(echo) + signal_processing.SignalProcessingUtils.SaveWav(mix_filepath, mix) + + # Check if hard clipping occurs. + if mix is None: + mix = signal_processing.SignalProcessingUtils.LoadWav(mix_filepath) + if signal_processing.SignalProcessingUtils.DetectHardClipping(mix): + logging.warning(cls._HARD_CLIPPING_LOG_MSG) + + return mix_filepath diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py new file mode 100644 index 0000000000..4fd5e4f1ee --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py @@ -0,0 +1,140 @@ +# Copyright (c) 2017 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. +"""Unit tests for the input mixer module. +""" + +import logging +import os +import shutil +import tempfile +import unittest + +import mock + +from . import exceptions +from . import input_mixer +from . import signal_processing + + +class TestApmInputMixer(unittest.TestCase): + """Unit tests for the ApmInputMixer class. + """ + + # Audio track file names created in setUp(). + _FILENAMES = ['capture', 'echo_1', 'echo_2', 'shorter', 'longer'] + + # Target peak power level (dBFS) of each audio track file created in setUp(). + # These values are hand-crafted in order to make saturation happen when + # capture and echo_2 are mixed and the contrary for capture and echo_1. + # None means that the power is not changed. + _MAX_PEAK_POWER_LEVELS = [-10.0, -5.0, 0.0, None, None] + + # Audio track file durations in milliseconds. + _DURATIONS = [1000, 1000, 1000, 800, 1200] + + _SAMPLE_RATE = 48000 + + def setUp(self): + """Creates temporary data.""" + self._tmp_path = tempfile.mkdtemp() + + # Create audio track files. + self._audio_tracks = {} + for filename, peak_power, duration in zip(self._FILENAMES, + self._MAX_PEAK_POWER_LEVELS, + self._DURATIONS): + audio_track_filepath = os.path.join(self._tmp_path, + '{}.wav'.format(filename)) + + # Create a pure tone with the target peak power level. + template = signal_processing.SignalProcessingUtils.GenerateSilence( + duration=duration, sample_rate=self._SAMPLE_RATE) + signal = signal_processing.SignalProcessingUtils.GeneratePureTone( + template) + if peak_power is not None: + signal = signal.apply_gain(-signal.max_dBFS + peak_power) + + signal_processing.SignalProcessingUtils.SaveWav( + audio_track_filepath, signal) + self._audio_tracks[filename] = { + 'filepath': + audio_track_filepath, + 'num_samples': + signal_processing.SignalProcessingUtils.CountSamples(signal) + } + + def tearDown(self): + """Recursively deletes temporary folders.""" + shutil.rmtree(self._tmp_path) + + def testCheckMixSameDuration(self): + """Checks the duration when mixing capture and echo with same duration.""" + mix_filepath = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['echo_1']['filepath']) + self.assertTrue(os.path.exists(mix_filepath)) + + mix = signal_processing.SignalProcessingUtils.LoadWav(mix_filepath) + self.assertEqual( + self._audio_tracks['capture']['num_samples'], + signal_processing.SignalProcessingUtils.CountSamples(mix)) + + def testRejectShorterEcho(self): + """Rejects echo signals that are shorter than the capture signal.""" + try: + _ = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['shorter']['filepath']) + self.fail('no exception raised') + except exceptions.InputMixerException: + pass + + def testCheckMixDurationWithLongerEcho(self): + """Checks the duration when mixing an echo longer than the capture.""" + mix_filepath = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['longer']['filepath']) + self.assertTrue(os.path.exists(mix_filepath)) + + mix = signal_processing.SignalProcessingUtils.LoadWav(mix_filepath) + self.assertEqual( + self._audio_tracks['capture']['num_samples'], + signal_processing.SignalProcessingUtils.CountSamples(mix)) + + def testCheckOutputFileNamesConflict(self): + """Checks that different echo files lead to different output file names.""" + mix1_filepath = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['echo_1']['filepath']) + self.assertTrue(os.path.exists(mix1_filepath)) + + mix2_filepath = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['echo_2']['filepath']) + self.assertTrue(os.path.exists(mix2_filepath)) + + self.assertNotEqual(mix1_filepath, mix2_filepath) + + def testHardClippingLogExpected(self): + """Checks that hard clipping warning is raised when occurring.""" + logging.warning = mock.MagicMock(name='warning') + _ = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['echo_2']['filepath']) + logging.warning.assert_called_once_with( + input_mixer.ApmInputMixer.HardClippingLogMessage()) + + def testHardClippingLogNotExpected(self): + """Checks that hard clipping warning is not raised when not occurring.""" + logging.warning = mock.MagicMock(name='warning') + _ = input_mixer.ApmInputMixer.Mix( + self._tmp_path, self._audio_tracks['capture']['filepath'], + self._audio_tracks['echo_1']['filepath']) + self.assertNotIn( + mock.call(input_mixer.ApmInputMixer.HardClippingLogMessage()), + logging.warning.call_args_list) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py new file mode 100644 index 0000000000..b64fdcca89 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py @@ -0,0 +1,68 @@ +# Copyright (c) 2017 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. +"""Input signal creator module. +""" + +from . import exceptions +from . import signal_processing + + +class InputSignalCreator(object): + """Input signal creator class. + """ + + @classmethod + def Create(cls, name, raw_params): + """Creates a input signal and its metadata. + + Args: + name: Input signal creator name. + raw_params: Tuple of parameters to pass to the specific signal creator. + + Returns: + (AudioSegment, dict) tuple. + """ + try: + signal = {} + params = {} + + if name == 'pure_tone': + params['frequency'] = float(raw_params[0]) + params['duration'] = int(raw_params[1]) + signal = cls._CreatePureTone(params['frequency'], + params['duration']) + else: + raise exceptions.InputSignalCreatorException( + 'Invalid input signal creator name') + + # Complete metadata. + params['signal'] = name + + return signal, params + except (TypeError, AssertionError) as e: + raise exceptions.InputSignalCreatorException( + 'Invalid signal creator parameters: {}'.format(e)) + + @classmethod + def _CreatePureTone(cls, frequency, duration): + """ + Generates a pure tone at 48000 Hz. + + Args: + frequency: Float in (0-24000] (Hz). + duration: Integer (milliseconds). + + Returns: + AudioSegment instance. + """ + assert 0 < frequency <= 24000 + assert duration > 0 + template = signal_processing.SignalProcessingUtils.GenerateSilence( + duration) + return signal_processing.SignalProcessingUtils.GeneratePureTone( + template, frequency) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css new file mode 100644 index 0000000000..2f406bb002 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css @@ -0,0 +1,32 @@ +/* Copyright (c) 2017 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. + */ + +td.selected-score { + background-color: #DDD; +} + +td.single-score-cell{ + text-align: center; +} + +.audio-inspector { + text-align: center; +} + +.audio-inspector div{ + margin-bottom: 0; + padding-bottom: 0; + padding-top: 0; +} + +.audio-inspector div div{ + margin-bottom: 0; + padding-bottom: 0; + padding-top: 0; +} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js new file mode 100644 index 0000000000..8e47411058 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js @@ -0,0 +1,376 @@ +// Copyright (c) 2017 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. + +/** + * Opens the score stats inspector dialog. + * @param {String} dialogId: identifier of the dialog to show. + * @return {DOMElement} The dialog element that has been opened. + */ +function openScoreStatsInspector(dialogId) { + var dialog = document.getElementById(dialogId); + dialog.showModal(); + return dialog; +} + +/** + * Closes the score stats inspector dialog. + */ +function closeScoreStatsInspector() { + var dialog = document.querySelector('dialog[open]'); + if (dialog == null) + return; + dialog.close(); +} + +/** + * Audio inspector class. + * @constructor + */ +function AudioInspector() { + console.debug('Creating an AudioInspector instance.'); + this.audioPlayer_ = new Audio(); + this.metadata_ = {}; + this.currentScore_ = null; + this.audioInspector_ = null; + this.snackbarContainer_ = document.querySelector('#snackbar'); + + // Get base URL without anchors. + this.baseUrl_ = window.location.href; + var index = this.baseUrl_.indexOf('#'); + if (index > 0) + this.baseUrl_ = this.baseUrl_.substr(0, index) + console.info('Base URL set to "' + window.location.href + '".'); + + window.event.stopPropagation(); + this.createTextAreasForCopy_(); + this.createAudioInspector_(); + this.initializeEventHandlers_(); + + // When MDL is ready, parse the anchor (if any) to show the requested + // experiment. + var self = this; + document.querySelectorAll('header a')[0].addEventListener( + 'mdl-componentupgraded', function() { + if (!self.parseWindowAnchor()) { + // If not experiment is requested, open the first section. + console.info('No anchor parsing, opening the first section.'); + document.querySelectorAll('header a > span')[0].click(); + } + }); +} + +/** + * Parse the anchor in the window URL. + * @return {bool} True if the parsing succeeded. + */ +AudioInspector.prototype.parseWindowAnchor = function() { + var index = location.href.indexOf('#'); + if (index == -1) { + console.debug('No # found in the URL.'); + return false; + } + + var anchor = location.href.substr(index - location.href.length + 1); + console.info('Anchor changed: "' + anchor + '".'); + + var parts = anchor.split('&'); + if (parts.length != 3) { + console.info('Ignoring anchor with invalid number of fields.'); + return false; + } + + var openDialog = document.querySelector('dialog[open]'); + try { + // Open the requested dialog if not already open. + if (!openDialog || openDialog.id != parts[1]) { + !openDialog || openDialog.close(); + document.querySelectorAll('header a > span')[ + parseInt(parts[0].substr(1))].click(); + openDialog = openScoreStatsInspector(parts[1]); + } + + // Trigger click on cell. + var cell = openDialog.querySelector('td.' + parts[2]); + cell.focus(); + cell.click(); + + this.showNotification_('Experiment selected.'); + return true; + } catch (e) { + this.showNotification_('Cannot select experiment :('); + console.error('Exception caught while selecting experiment: "' + e + '".'); + } + + return false; +} + +/** + * Set up the inspector for a new score. + * @param {DOMElement} element: Element linked to the selected score. + */ +AudioInspector.prototype.selectedScoreChange = function(element) { + if (this.currentScore_ == element) { return; } + if (this.currentScore_ != null) { + this.currentScore_.classList.remove('selected-score'); + } + this.currentScore_ = element; + this.currentScore_.classList.add('selected-score'); + this.stopAudio(); + + // Read metadata. + var matches = element.querySelectorAll('input[type=hidden]'); + this.metadata_ = {}; + for (var index = 0; index < matches.length; ++index) { + this.metadata_[matches[index].name] = matches[index].value; + } + + // Show the audio inspector interface. + var container = element.parentNode.parentNode.parentNode.parentNode; + var audioInspectorPlaceholder = container.querySelector( + '.audio-inspector-placeholder'); + this.moveInspector_(audioInspectorPlaceholder); +}; + +/** + * Stop playing audio. + */ +AudioInspector.prototype.stopAudio = function() { + console.info('Pausing audio play out.'); + this.audioPlayer_.pause(); +}; + +/** + * Show a text message using the snackbar. + */ +AudioInspector.prototype.showNotification_ = function(text) { + try { + this.snackbarContainer_.MaterialSnackbar.showSnackbar({ + message: text, timeout: 2000}); + } catch (e) { + // Fallback to an alert. + alert(text); + console.warn('Cannot use snackbar: "' + e + '"'); + } +} + +/** + * Move the audio inspector DOM node into the given parent. + * @param {DOMElement} newParentNode: New parent for the inspector. + */ +AudioInspector.prototype.moveInspector_ = function(newParentNode) { + newParentNode.appendChild(this.audioInspector_); +}; + +/** + * Play audio file from url. + * @param {string} metadataFieldName: Metadata field name. + */ +AudioInspector.prototype.playAudio = function(metadataFieldName) { + if (this.metadata_[metadataFieldName] == undefined) { return; } + if (this.metadata_[metadataFieldName] == 'None') { + alert('The selected stream was not used during the experiment.'); + return; + } + this.stopAudio(); + this.audioPlayer_.src = this.metadata_[metadataFieldName]; + console.debug('Audio source URL: "' + this.audioPlayer_.src + '"'); + this.audioPlayer_.play(); + console.info('Playing out audio.'); +}; + +/** + * Create hidden text areas to copy URLs. + * + * For each dialog, one text area is created since it is not possible to select + * text on a text area outside of the active dialog. + */ +AudioInspector.prototype.createTextAreasForCopy_ = function() { + var self = this; + document.querySelectorAll('dialog.mdl-dialog').forEach(function(element) { + var textArea = document.createElement("textarea"); + textArea.classList.add('url-copy'); + textArea.style.position = 'fixed'; + textArea.style.bottom = 0; + textArea.style.left = 0; + textArea.style.width = '2em'; + textArea.style.height = '2em'; + textArea.style.border = 'none'; + textArea.style.outline = 'none'; + textArea.style.boxShadow = 'none'; + textArea.style.background = 'transparent'; + textArea.style.fontSize = '6px'; + element.appendChild(textArea); + }); +} + +/** + * Create audio inspector. + */ +AudioInspector.prototype.createAudioInspector_ = function() { + var buttonIndex = 0; + function getButtonHtml(icon, toolTipText, caption, metadataFieldName) { + var buttonId = 'audioInspectorButton' + buttonIndex++; + html = caption == null ? '' : caption; + html += '' + + return html; + } + + // TODO(alessiob): Add timeline and highlight current track by changing icon + // color. + + this.audioInspector_ = document.createElement('div'); + this.audioInspector_.classList.add('audio-inspector'); + this.audioInspector_.innerHTML = + '
' + + '
' + + '
' + + getButtonHtml('play_arrow', 'Simulated echo', 'Ein', + 'echo_filepath') + + '
' + + '
' + + getButtonHtml('stop', 'Stop playing [S]', null, '__stop__') + + '
' + + '
' + + getButtonHtml('play_arrow', 'Render stream', 'Rin', + 'render_filepath') + + '
' + + '
' + + '
' + + '
' + + '
' + + '
' + + getButtonHtml('play_arrow', 'Capture stream (APM input) [1]', + 'Y\'in', 'capture_filepath') + + '
' + + '
APM
' + + '
' + + getButtonHtml('play_arrow', 'APM output [2]', 'Yout', + 'apm_output_filepath') + + '
' + + '
' + + '
' + + '
' + + '
' + + '
' + + getButtonHtml('play_arrow', 'Echo-free capture stream', + 'Yin', 'echo_free_capture_filepath') + + '
' + + '
' + + getButtonHtml('play_arrow', 'Clean capture stream', + 'Yclean', 'clean_capture_input_filepath') + + '
' + + '
' + + getButtonHtml('play_arrow', 'APM reference [3]', 'Yref', + 'apm_reference_filepath') + + '
' + + '
' + + '
'; + + // Add an invisible node as initial container for the audio inspector. + var parent = document.createElement('div'); + parent.style.display = 'none'; + this.moveInspector_(parent); + document.body.appendChild(parent); +}; + +/** + * Initialize event handlers. + */ +AudioInspector.prototype.initializeEventHandlers_ = function() { + var self = this; + + // Score cells. + document.querySelectorAll('td.single-score-cell').forEach(function(element) { + element.onclick = function() { + self.selectedScoreChange(this); + } + }); + + // Copy anchor URLs icons. + if (document.queryCommandSupported('copy')) { + document.querySelectorAll('td.single-score-cell button').forEach( + function(element) { + element.onclick = function() { + // Find the text area in the dialog. + var textArea = element.closest('dialog').querySelector( + 'textarea.url-copy'); + + // Copy. + textArea.value = self.baseUrl_ + '#' + element.getAttribute( + 'data-anchor'); + textArea.select(); + try { + if (!document.execCommand('copy')) + throw 'Copy returned false'; + self.showNotification_('Experiment URL copied.'); + } catch (e) { + self.showNotification_('Cannot copy experiment URL :('); + console.error(e); + } + } + }); + } else { + self.showNotification_( + 'The copy command is disabled. URL copy is not enabled.'); + } + + // Audio inspector buttons. + this.audioInspector_.querySelectorAll('button').forEach(function(element) { + var target = element.querySelector('input[type=hidden]'); + if (target == null) { return; } + element.onclick = function() { + if (target.value == '__stop__') { + self.stopAudio(); + } else { + self.playAudio(target.value); + } + }; + }); + + // Dialog close handlers. + var dialogs = document.querySelectorAll('dialog').forEach(function(element) { + element.onclose = function() { + self.stopAudio(); + } + }); + + // Keyboard shortcuts. + window.onkeyup = function(e) { + var key = e.keyCode ? e.keyCode : e.which; + switch (key) { + case 49: // 1. + self.playAudio('capture_filepath'); + break; + case 50: // 2. + self.playAudio('apm_output_filepath'); + break; + case 51: // 3. + self.playAudio('apm_reference_filepath'); + break; + case 83: // S. + case 115: // s. + self.stopAudio(); + break; + } + }; + + // Hash change. + window.onhashchange = function(e) { + self.parseWindowAnchor(); + } +}; diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py new file mode 100644 index 0000000000..95e801903d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py @@ -0,0 +1,359 @@ +# Copyright (c) 2017 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. +"""Signal processing utility module. +""" + +import array +import logging +import os +import sys +import enum + +try: + import numpy as np +except ImportError: + logging.critical('Cannot import the third-party Python package numpy') + sys.exit(1) + +try: + import pydub + import pydub.generators +except ImportError: + logging.critical('Cannot import the third-party Python package pydub') + sys.exit(1) + +try: + import scipy.signal + import scipy.fftpack +except ImportError: + logging.critical('Cannot import the third-party Python package scipy') + sys.exit(1) + +from . import exceptions + + +class SignalProcessingUtils(object): + """Collection of signal processing utilities. + """ + + @enum.unique + class MixPadding(enum.Enum): + NO_PADDING = 0 + ZERO_PADDING = 1 + LOOP = 2 + + def __init__(self): + pass + + @classmethod + def LoadWav(cls, filepath, channels=1): + """Loads wav file. + + Args: + filepath: path to the wav audio track file to load. + channels: number of channels (downmixing to mono by default). + + Returns: + AudioSegment instance. + """ + if not os.path.exists(filepath): + logging.error('cannot find the <%s> audio track file', filepath) + raise exceptions.FileNotFoundError() + return pydub.AudioSegment.from_file(filepath, + format='wav', + channels=channels) + + @classmethod + def SaveWav(cls, output_filepath, signal): + """Saves wav file. + + Args: + output_filepath: path to the wav audio track file to save. + signal: AudioSegment instance. + """ + return signal.export(output_filepath, format='wav') + + @classmethod + def CountSamples(cls, signal): + """Number of samples per channel. + + Args: + signal: AudioSegment instance. + + Returns: + An integer. + """ + number_of_samples = len(signal.get_array_of_samples()) + assert signal.channels > 0 + assert number_of_samples % signal.channels == 0 + return number_of_samples / signal.channels + + @classmethod + def GenerateSilence(cls, duration=1000, sample_rate=48000): + """Generates silence. + + This method can also be used to create a template AudioSegment instance. + A template can then be used with other Generate*() methods accepting an + AudioSegment instance as argument. + + Args: + duration: duration in ms. + sample_rate: sample rate. + + Returns: + AudioSegment instance. + """ + return pydub.AudioSegment.silent(duration, sample_rate) + + @classmethod + def GeneratePureTone(cls, template, frequency=440.0): + """Generates a pure tone. + + The pure tone is generated with the same duration and in the same format of + the given template signal. + + Args: + template: AudioSegment instance. + frequency: Frequency of the pure tone in Hz. + + Return: + AudioSegment instance. + """ + if frequency > template.frame_rate >> 1: + raise exceptions.SignalProcessingException('Invalid frequency') + + generator = pydub.generators.Sine(sample_rate=template.frame_rate, + bit_depth=template.sample_width * 8, + freq=frequency) + + return generator.to_audio_segment(duration=len(template), volume=0.0) + + @classmethod + def GenerateWhiteNoise(cls, template): + """Generates white noise. + + The white noise is generated with the same duration and in the same format + of the given template signal. + + Args: + template: AudioSegment instance. + + Return: + AudioSegment instance. + """ + generator = pydub.generators.WhiteNoise( + sample_rate=template.frame_rate, + bit_depth=template.sample_width * 8) + return generator.to_audio_segment(duration=len(template), volume=0.0) + + @classmethod + def AudioSegmentToRawData(cls, signal): + samples = signal.get_array_of_samples() + if samples.typecode != 'h': + raise exceptions.SignalProcessingException( + 'Unsupported samples type') + return np.array(signal.get_array_of_samples(), np.int16) + + @classmethod + def Fft(cls, signal, normalize=True): + if signal.channels != 1: + raise NotImplementedError('multiple-channel FFT not implemented') + x = cls.AudioSegmentToRawData(signal).astype(np.float32) + if normalize: + x /= max(abs(np.max(x)), 1.0) + y = scipy.fftpack.fft(x) + return y[:len(y) / 2] + + @classmethod + def DetectHardClipping(cls, signal, threshold=2): + """Detects hard clipping. + + Hard clipping is simply detected by counting samples that touch either the + lower or upper bound too many times in a row (according to `threshold`). + The presence of a single sequence of samples meeting such property is enough + to label the signal as hard clipped. + + Args: + signal: AudioSegment instance. + threshold: minimum number of samples at full-scale in a row. + + Returns: + True if hard clipping is detect, False otherwise. + """ + if signal.channels != 1: + raise NotImplementedError( + 'multiple-channel clipping not implemented') + if signal.sample_width != 2: # Note that signal.sample_width is in bytes. + raise exceptions.SignalProcessingException( + 'hard-clipping detection only supported for 16 bit samples') + samples = cls.AudioSegmentToRawData(signal) + + # Detect adjacent clipped samples. + samples_type_info = np.iinfo(samples.dtype) + mask_min = samples == samples_type_info.min + mask_max = samples == samples_type_info.max + + def HasLongSequence(vector, min_legth=threshold): + """Returns True if there are one or more long sequences of True flags.""" + seq_length = 0 + for b in vector: + seq_length = seq_length + 1 if b else 0 + if seq_length >= min_legth: + return True + return False + + return HasLongSequence(mask_min) or HasLongSequence(mask_max) + + @classmethod + def ApplyImpulseResponse(cls, signal, impulse_response): + """Applies an impulse response to a signal. + + Args: + signal: AudioSegment instance. + impulse_response: list or numpy vector of float values. + + Returns: + AudioSegment instance. + """ + # Get samples. + assert signal.channels == 1, ( + 'multiple-channel recordings not supported') + samples = signal.get_array_of_samples() + + # Convolve. + logging.info( + 'applying %d order impulse response to a signal lasting %d ms', + len(impulse_response), len(signal)) + convolved_samples = scipy.signal.fftconvolve(in1=samples, + in2=impulse_response, + mode='full').astype( + np.int16) + logging.info('convolution computed') + + # Cast. + convolved_samples = array.array(signal.array_type, convolved_samples) + + # Verify. + logging.debug('signal length: %d samples', len(samples)) + logging.debug('convolved signal length: %d samples', + len(convolved_samples)) + assert len(convolved_samples) > len(samples) + + # Generate convolved signal AudioSegment instance. + convolved_signal = pydub.AudioSegment(data=convolved_samples, + metadata={ + 'sample_width': + signal.sample_width, + 'frame_rate': + signal.frame_rate, + 'frame_width': + signal.frame_width, + 'channels': signal.channels, + }) + assert len(convolved_signal) > len(signal) + + return convolved_signal + + @classmethod + def Normalize(cls, signal): + """Normalizes a signal. + + Args: + signal: AudioSegment instance. + + Returns: + An AudioSegment instance. + """ + return signal.apply_gain(-signal.max_dBFS) + + @classmethod + def Copy(cls, signal): + """Makes a copy os a signal. + + Args: + signal: AudioSegment instance. + + Returns: + An AudioSegment instance. + """ + return pydub.AudioSegment(data=signal.get_array_of_samples(), + metadata={ + 'sample_width': signal.sample_width, + 'frame_rate': signal.frame_rate, + 'frame_width': signal.frame_width, + 'channels': signal.channels, + }) + + @classmethod + def MixSignals(cls, + signal, + noise, + target_snr=0.0, + pad_noise=MixPadding.NO_PADDING): + """Mixes `signal` and `noise` with a target SNR. + + Mix `signal` and `noise` with a desired SNR by scaling `noise`. + If the target SNR is +/- infinite, a copy of signal/noise is returned. + If `signal` is shorter than `noise`, the length of the mix equals that of + `signal`. Otherwise, the mix length depends on whether padding is applied. + When padding is not applied, that is `pad_noise` is set to NO_PADDING + (default), the mix length equals that of `noise` - i.e., `signal` is + truncated. Otherwise, `noise` is extended and the resulting mix has the same + length of `signal`. + + Args: + signal: AudioSegment instance (signal). + noise: AudioSegment instance (noise). + target_snr: float, numpy.Inf or -numpy.Inf (dB). + pad_noise: SignalProcessingUtils.MixPadding, default: NO_PADDING. + + Returns: + An AudioSegment instance. + """ + # Handle infinite target SNR. + if target_snr == -np.Inf: + # Return a copy of noise. + logging.warning('SNR = -Inf, returning noise') + return cls.Copy(noise) + elif target_snr == np.Inf: + # Return a copy of signal. + logging.warning('SNR = +Inf, returning signal') + return cls.Copy(signal) + + # Check signal and noise power. + signal_power = float(signal.dBFS) + noise_power = float(noise.dBFS) + if signal_power == -np.Inf: + logging.error('signal has -Inf power, cannot mix') + raise exceptions.SignalProcessingException( + 'cannot mix a signal with -Inf power') + if noise_power == -np.Inf: + logging.error('noise has -Inf power, cannot mix') + raise exceptions.SignalProcessingException( + 'cannot mix a signal with -Inf power') + + # Mix. + gain_db = signal_power - noise_power - target_snr + signal_duration = len(signal) + noise_duration = len(noise) + if signal_duration <= noise_duration: + # Ignore `pad_noise`, `noise` is truncated if longer that `signal`, the + # mix will have the same length of `signal`. + return signal.overlay(noise.apply_gain(gain_db)) + elif pad_noise == cls.MixPadding.NO_PADDING: + # `signal` is longer than `noise`, but no padding is applied to `noise`. + # Truncate `signal`. + return noise.overlay(signal, gain_during_overlay=gain_db) + elif pad_noise == cls.MixPadding.ZERO_PADDING: + # TODO(alessiob): Check that this works as expected. + return signal.overlay(noise.apply_gain(gain_db)) + elif pad_noise == cls.MixPadding.LOOP: + # `signal` is longer than `noise`, extend `noise` by looping. + return signal.overlay(noise.apply_gain(gain_db), loop=True) + else: + raise exceptions.SignalProcessingException('invalid padding type') diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py new file mode 100644 index 0000000000..881fb66800 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py @@ -0,0 +1,183 @@ +# Copyright (c) 2017 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. +"""Unit tests for the signal_processing module. +""" + +import unittest + +import numpy as np +import pydub + +from . import exceptions +from . import signal_processing + + +class TestSignalProcessing(unittest.TestCase): + """Unit tests for the signal_processing module. + """ + + def testMixSignals(self): + # Generate a template signal with which white noise can be generated. + silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000) + + # Generate two distinct AudioSegment instances with 1 second of white noise. + signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + silence) + noise = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + silence) + + # Extract samples. + signal_samples = signal.get_array_of_samples() + noise_samples = noise.get_array_of_samples() + + # Test target SNR -Inf (noise expected). + mix_neg_inf = signal_processing.SignalProcessingUtils.MixSignals( + signal, noise, -np.Inf) + self.assertTrue(len(noise), len(mix_neg_inf)) # Check duration. + mix_neg_inf_samples = mix_neg_inf.get_array_of_samples() + self.assertTrue( # Check samples. + all([x == y for x, y in zip(noise_samples, mix_neg_inf_samples)])) + + # Test target SNR 0.0 (different data expected). + mix_0 = signal_processing.SignalProcessingUtils.MixSignals( + signal, noise, 0.0) + self.assertTrue(len(signal), len(mix_0)) # Check duration. + self.assertTrue(len(noise), len(mix_0)) + mix_0_samples = mix_0.get_array_of_samples() + self.assertTrue( + any([x != y for x, y in zip(signal_samples, mix_0_samples)])) + self.assertTrue( + any([x != y for x, y in zip(noise_samples, mix_0_samples)])) + + # Test target SNR +Inf (signal expected). + mix_pos_inf = signal_processing.SignalProcessingUtils.MixSignals( + signal, noise, np.Inf) + self.assertTrue(len(signal), len(mix_pos_inf)) # Check duration. + mix_pos_inf_samples = mix_pos_inf.get_array_of_samples() + self.assertTrue( # Check samples. + all([x == y for x, y in zip(signal_samples, mix_pos_inf_samples)])) + + def testMixSignalsMinInfPower(self): + silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000) + signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + silence) + + with self.assertRaises(exceptions.SignalProcessingException): + _ = signal_processing.SignalProcessingUtils.MixSignals( + signal, silence, 0.0) + + with self.assertRaises(exceptions.SignalProcessingException): + _ = signal_processing.SignalProcessingUtils.MixSignals( + silence, signal, 0.0) + + def testMixSignalNoiseDifferentLengths(self): + # Test signals. + shorter = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + pydub.AudioSegment.silent(duration=1000, frame_rate=8000)) + longer = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + pydub.AudioSegment.silent(duration=2000, frame_rate=8000)) + + # When the signal is shorter than the noise, the mix length always equals + # that of the signal regardless of whether padding is applied. + # No noise padding, length of signal less than that of noise. + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=shorter, + noise=longer, + pad_noise=signal_processing.SignalProcessingUtils.MixPadding. + NO_PADDING) + self.assertEqual(len(shorter), len(mix)) + # With noise padding, length of signal less than that of noise. + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=shorter, + noise=longer, + pad_noise=signal_processing.SignalProcessingUtils.MixPadding. + ZERO_PADDING) + self.assertEqual(len(shorter), len(mix)) + + # When the signal is longer than the noise, the mix length depends on + # whether padding is applied. + # No noise padding, length of signal greater than that of noise. + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=longer, + noise=shorter, + pad_noise=signal_processing.SignalProcessingUtils.MixPadding. + NO_PADDING) + self.assertEqual(len(shorter), len(mix)) + # With noise padding, length of signal greater than that of noise. + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=longer, + noise=shorter, + pad_noise=signal_processing.SignalProcessingUtils.MixPadding. + ZERO_PADDING) + self.assertEqual(len(longer), len(mix)) + + def testMixSignalNoisePaddingTypes(self): + # Test signals. + shorter = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + pydub.AudioSegment.silent(duration=1000, frame_rate=8000)) + longer = signal_processing.SignalProcessingUtils.GeneratePureTone( + pydub.AudioSegment.silent(duration=2000, frame_rate=8000), 440.0) + + # Zero padding: expect pure tone only in 1-2s. + mix_zero_pad = signal_processing.SignalProcessingUtils.MixSignals( + signal=longer, + noise=shorter, + target_snr=-6, + pad_noise=signal_processing.SignalProcessingUtils.MixPadding. + ZERO_PADDING) + + # Loop: expect pure tone plus noise in 1-2s. + mix_loop = signal_processing.SignalProcessingUtils.MixSignals( + signal=longer, + noise=shorter, + target_snr=-6, + pad_noise=signal_processing.SignalProcessingUtils.MixPadding.LOOP) + + def Energy(signal): + samples = signal_processing.SignalProcessingUtils.AudioSegmentToRawData( + signal).astype(np.float32) + return np.sum(samples * samples) + + e_mix_zero_pad = Energy(mix_zero_pad[-1000:]) + e_mix_loop = Energy(mix_loop[-1000:]) + self.assertLess(0, e_mix_zero_pad) + self.assertLess(e_mix_zero_pad, e_mix_loop) + + def testMixSignalSnr(self): + # Test signals. + tone_low = signal_processing.SignalProcessingUtils.GeneratePureTone( + pydub.AudioSegment.silent(duration=64, frame_rate=8000), 250.0) + tone_high = signal_processing.SignalProcessingUtils.GeneratePureTone( + pydub.AudioSegment.silent(duration=64, frame_rate=8000), 3000.0) + + def ToneAmplitudes(mix): + """Returns the amplitude of the coefficients #16 and #192, which + correspond to the tones at 250 and 3k Hz respectively.""" + mix_fft = np.absolute( + signal_processing.SignalProcessingUtils.Fft(mix)) + return mix_fft[16], mix_fft[192] + + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=tone_low, noise=tone_high, target_snr=-6) + ampl_low, ampl_high = ToneAmplitudes(mix) + self.assertLess(ampl_low, ampl_high) + + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=tone_high, noise=tone_low, target_snr=-6) + ampl_low, ampl_high = ToneAmplitudes(mix) + self.assertLess(ampl_high, ampl_low) + + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=tone_low, noise=tone_high, target_snr=6) + ampl_low, ampl_high = ToneAmplitudes(mix) + self.assertLess(ampl_high, ampl_low) + + mix = signal_processing.SignalProcessingUtils.MixSignals( + signal=tone_high, noise=tone_low, target_snr=6) + ampl_low, ampl_high = ToneAmplitudes(mix) + self.assertLess(ampl_low, ampl_high) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py new file mode 100644 index 0000000000..69b3a1624e --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py @@ -0,0 +1,446 @@ +# Copyright (c) 2017 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. +"""APM module simulator. +""" + +import logging +import os + +from . import annotations +from . import data_access +from . import echo_path_simulation +from . import echo_path_simulation_factory +from . import eval_scores +from . import exceptions +from . import input_mixer +from . import input_signal_creator +from . import signal_processing +from . import test_data_generation + + +class ApmModuleSimulator(object): + """Audio processing module (APM) simulator class. + """ + + _TEST_DATA_GENERATOR_CLASSES = ( + test_data_generation.TestDataGenerator.REGISTERED_CLASSES) + _EVAL_SCORE_WORKER_CLASSES = eval_scores.EvaluationScore.REGISTERED_CLASSES + + _PREFIX_APM_CONFIG = 'apmcfg-' + _PREFIX_CAPTURE = 'capture-' + _PREFIX_RENDER = 'render-' + _PREFIX_ECHO_SIMULATOR = 'echosim-' + _PREFIX_TEST_DATA_GEN = 'datagen-' + _PREFIX_TEST_DATA_GEN_PARAMS = 'datagen_params-' + _PREFIX_SCORE = 'score-' + + def __init__(self, + test_data_generator_factory, + evaluation_score_factory, + ap_wrapper, + evaluator, + external_vads=None): + if external_vads is None: + external_vads = {} + self._test_data_generator_factory = test_data_generator_factory + self._evaluation_score_factory = evaluation_score_factory + self._audioproc_wrapper = ap_wrapper + self._evaluator = evaluator + self._annotator = annotations.AudioAnnotationsExtractor( + annotations.AudioAnnotationsExtractor.VadType.ENERGY_THRESHOLD + | annotations.AudioAnnotationsExtractor.VadType.WEBRTC_COMMON_AUDIO + | annotations.AudioAnnotationsExtractor.VadType.WEBRTC_APM, + external_vads) + + # Init. + self._test_data_generator_factory.SetOutputDirectoryPrefix( + self._PREFIX_TEST_DATA_GEN_PARAMS) + self._evaluation_score_factory.SetScoreFilenamePrefix( + self._PREFIX_SCORE) + + # Properties for each run. + self._base_output_path = None + self._output_cache_path = None + self._test_data_generators = None + self._evaluation_score_workers = None + self._config_filepaths = None + self._capture_input_filepaths = None + self._render_input_filepaths = None + self._echo_path_simulator_class = None + + @classmethod + def GetPrefixApmConfig(cls): + return cls._PREFIX_APM_CONFIG + + @classmethod + def GetPrefixCapture(cls): + return cls._PREFIX_CAPTURE + + @classmethod + def GetPrefixRender(cls): + return cls._PREFIX_RENDER + + @classmethod + def GetPrefixEchoSimulator(cls): + return cls._PREFIX_ECHO_SIMULATOR + + @classmethod + def GetPrefixTestDataGenerator(cls): + return cls._PREFIX_TEST_DATA_GEN + + @classmethod + def GetPrefixTestDataGeneratorParameters(cls): + return cls._PREFIX_TEST_DATA_GEN_PARAMS + + @classmethod + def GetPrefixScore(cls): + return cls._PREFIX_SCORE + + def Run(self, + config_filepaths, + capture_input_filepaths, + test_data_generator_names, + eval_score_names, + output_dir, + render_input_filepaths=None, + echo_path_simulator_name=( + echo_path_simulation.NoEchoPathSimulator.NAME)): + """Runs the APM simulation. + + Initializes paths and required instances, then runs all the simulations. + The render input can be optionally added. If added, the number of capture + input audio tracks and the number of render input audio tracks have to be + equal. The two lists are used to form pairs of capture and render input. + + Args: + config_filepaths: set of APM configuration files to test. + capture_input_filepaths: set of capture input audio track files to test. + test_data_generator_names: set of test data generator names to test. + eval_score_names: set of evaluation score names to test. + output_dir: base path to the output directory for wav files and outcomes. + render_input_filepaths: set of render input audio track files to test. + echo_path_simulator_name: name of the echo path simulator to use when + render input is provided. + """ + assert render_input_filepaths is None or ( + len(capture_input_filepaths) == len(render_input_filepaths)), ( + 'render input set size not matching input set size') + assert render_input_filepaths is None or echo_path_simulator_name in ( + echo_path_simulation.EchoPathSimulator.REGISTERED_CLASSES), ( + 'invalid echo path simulator') + self._base_output_path = os.path.abspath(output_dir) + + # Output path used to cache the data shared across simulations. + self._output_cache_path = os.path.join(self._base_output_path, + '_cache') + + # Instance test data generators. + self._test_data_generators = [ + self._test_data_generator_factory.GetInstance( + test_data_generators_class=( + self._TEST_DATA_GENERATOR_CLASSES[name])) + for name in (test_data_generator_names) + ] + + # Instance evaluation score workers. + self._evaluation_score_workers = [ + self._evaluation_score_factory.GetInstance( + evaluation_score_class=self._EVAL_SCORE_WORKER_CLASSES[name]) + for (name) in eval_score_names + ] + + # Set APM configuration file paths. + self._config_filepaths = self._CreatePathsCollection(config_filepaths) + + # Set probing signal file paths. + if render_input_filepaths is None: + # Capture input only. + self._capture_input_filepaths = self._CreatePathsCollection( + capture_input_filepaths) + self._render_input_filepaths = None + else: + # Set both capture and render input signals. + self._SetTestInputSignalFilePaths(capture_input_filepaths, + render_input_filepaths) + + # Set the echo path simulator class. + self._echo_path_simulator_class = ( + echo_path_simulation.EchoPathSimulator. + REGISTERED_CLASSES[echo_path_simulator_name]) + + self._SimulateAll() + + def _SimulateAll(self): + """Runs all the simulations. + + Iterates over the combinations of APM configurations, probing signals, and + test data generators. This method is mainly responsible for the creation of + the cache and output directories required in order to call _Simulate(). + """ + without_render_input = self._render_input_filepaths is None + + # Try different APM config files. + for config_name in self._config_filepaths: + config_filepath = self._config_filepaths[config_name] + + # Try different capture-render pairs. + for capture_input_name in self._capture_input_filepaths: + # Output path for the capture signal annotations. + capture_annotations_cache_path = os.path.join( + self._output_cache_path, + self._PREFIX_CAPTURE + capture_input_name) + data_access.MakeDirectory(capture_annotations_cache_path) + + # Capture. + capture_input_filepath = self._capture_input_filepaths[ + capture_input_name] + if not os.path.exists(capture_input_filepath): + # If the input signal file does not exist, try to create using the + # available input signal creators. + self._CreateInputSignal(capture_input_filepath) + assert os.path.exists(capture_input_filepath) + self._ExtractCaptureAnnotations( + capture_input_filepath, capture_annotations_cache_path) + + # Render and simulated echo path (optional). + render_input_filepath = None if without_render_input else ( + self._render_input_filepaths[capture_input_name]) + render_input_name = '(none)' if without_render_input else ( + self._ExtractFileName(render_input_filepath)) + echo_path_simulator = (echo_path_simulation_factory. + EchoPathSimulatorFactory.GetInstance( + self._echo_path_simulator_class, + render_input_filepath)) + + # Try different test data generators. + for test_data_generators in self._test_data_generators: + logging.info( + 'APM config preset: <%s>, capture: <%s>, render: <%s>,' + 'test data generator: <%s>, echo simulator: <%s>', + config_name, capture_input_name, render_input_name, + test_data_generators.NAME, echo_path_simulator.NAME) + + # Output path for the generated test data. + test_data_cache_path = os.path.join( + capture_annotations_cache_path, + self._PREFIX_TEST_DATA_GEN + test_data_generators.NAME) + data_access.MakeDirectory(test_data_cache_path) + logging.debug('test data cache path: <%s>', + test_data_cache_path) + + # Output path for the echo simulator and APM input mixer output. + echo_test_data_cache_path = os.path.join( + test_data_cache_path, + 'echosim-{}'.format(echo_path_simulator.NAME)) + data_access.MakeDirectory(echo_test_data_cache_path) + logging.debug('echo test data cache path: <%s>', + echo_test_data_cache_path) + + # Full output path. + output_path = os.path.join( + self._base_output_path, + self._PREFIX_APM_CONFIG + config_name, + self._PREFIX_CAPTURE + capture_input_name, + self._PREFIX_RENDER + render_input_name, + self._PREFIX_ECHO_SIMULATOR + echo_path_simulator.NAME, + self._PREFIX_TEST_DATA_GEN + test_data_generators.NAME) + data_access.MakeDirectory(output_path) + logging.debug('output path: <%s>', output_path) + + self._Simulate(test_data_generators, + capture_input_filepath, + render_input_filepath, test_data_cache_path, + echo_test_data_cache_path, output_path, + config_filepath, echo_path_simulator) + + @staticmethod + def _CreateInputSignal(input_signal_filepath): + """Creates a missing input signal file. + + The file name is parsed to extract input signal creator and params. If a + creator is matched and the parameters are valid, a new signal is generated + and written in `input_signal_filepath`. + + Args: + input_signal_filepath: Path to the input signal audio file to write. + + Raises: + InputSignalCreatorException + """ + filename = os.path.splitext( + os.path.split(input_signal_filepath)[-1])[0] + filename_parts = filename.split('-') + + if len(filename_parts) < 2: + raise exceptions.InputSignalCreatorException( + 'Cannot parse input signal file name') + + signal, metadata = input_signal_creator.InputSignalCreator.Create( + filename_parts[0], filename_parts[1].split('_')) + + signal_processing.SignalProcessingUtils.SaveWav( + input_signal_filepath, signal) + data_access.Metadata.SaveFileMetadata(input_signal_filepath, metadata) + + def _ExtractCaptureAnnotations(self, + input_filepath, + output_path, + annotation_name=""): + self._annotator.Extract(input_filepath) + self._annotator.Save(output_path, annotation_name) + + def _Simulate(self, test_data_generators, clean_capture_input_filepath, + render_input_filepath, test_data_cache_path, + echo_test_data_cache_path, output_path, config_filepath, + echo_path_simulator): + """Runs a single set of simulation. + + Simulates a given combination of APM configuration, probing signal, and + test data generator. It iterates over the test data generator + internal configurations. + + Args: + test_data_generators: TestDataGenerator instance. + clean_capture_input_filepath: capture input audio track file to be + processed by a test data generator and + not affected by echo. + render_input_filepath: render input audio track file to test. + test_data_cache_path: path for the generated test audio track files. + echo_test_data_cache_path: path for the echo simulator. + output_path: base output path for the test data generator. + config_filepath: APM configuration file to test. + echo_path_simulator: EchoPathSimulator instance. + """ + # Generate pairs of noisy input and reference signal files. + test_data_generators.Generate( + input_signal_filepath=clean_capture_input_filepath, + test_data_cache_path=test_data_cache_path, + base_output_path=output_path) + + # Extract metadata linked to the clean input file (if any). + apm_input_metadata = None + try: + apm_input_metadata = data_access.Metadata.LoadFileMetadata( + clean_capture_input_filepath) + except IOError as e: + apm_input_metadata = {} + apm_input_metadata['test_data_gen_name'] = test_data_generators.NAME + apm_input_metadata['test_data_gen_config'] = None + + # For each test data pair, simulate a call and evaluate. + for config_name in test_data_generators.config_names: + logging.info(' - test data generator config: <%s>', config_name) + apm_input_metadata['test_data_gen_config'] = config_name + + # Paths to the test data generator output. + # Note that the reference signal does not depend on the render input + # which is optional. + noisy_capture_input_filepath = ( + test_data_generators.noisy_signal_filepaths[config_name]) + reference_signal_filepath = ( + test_data_generators.reference_signal_filepaths[config_name]) + + # Output path for the evaluation (e.g., APM output file). + evaluation_output_path = test_data_generators.apm_output_paths[ + config_name] + + # Paths to the APM input signals. + echo_path_filepath = echo_path_simulator.Simulate( + echo_test_data_cache_path) + apm_input_filepath = input_mixer.ApmInputMixer.Mix( + echo_test_data_cache_path, noisy_capture_input_filepath, + echo_path_filepath) + + # Extract annotations for the APM input mix. + apm_input_basepath, apm_input_filename = os.path.split( + apm_input_filepath) + self._ExtractCaptureAnnotations( + apm_input_filepath, apm_input_basepath, + os.path.splitext(apm_input_filename)[0] + '-') + + # Simulate a call using APM. + self._audioproc_wrapper.Run( + config_filepath=config_filepath, + capture_input_filepath=apm_input_filepath, + render_input_filepath=render_input_filepath, + output_path=evaluation_output_path) + + try: + # Evaluate. + self._evaluator.Run( + evaluation_score_workers=self._evaluation_score_workers, + apm_input_metadata=apm_input_metadata, + apm_output_filepath=self._audioproc_wrapper. + output_filepath, + reference_input_filepath=reference_signal_filepath, + render_input_filepath=render_input_filepath, + output_path=evaluation_output_path, + ) + + # Save simulation metadata. + data_access.Metadata.SaveAudioTestDataPaths( + output_path=evaluation_output_path, + clean_capture_input_filepath=clean_capture_input_filepath, + echo_free_capture_filepath=noisy_capture_input_filepath, + echo_filepath=echo_path_filepath, + render_filepath=render_input_filepath, + capture_filepath=apm_input_filepath, + apm_output_filepath=self._audioproc_wrapper. + output_filepath, + apm_reference_filepath=reference_signal_filepath, + apm_config_filepath=config_filepath, + ) + except exceptions.EvaluationScoreException as e: + logging.warning('the evaluation failed: %s', e.message) + continue + + def _SetTestInputSignalFilePaths(self, capture_input_filepaths, + render_input_filepaths): + """Sets input and render input file paths collections. + + Pairs the input and render input files by storing the file paths into two + collections. The key is the file name of the input file. + + Args: + capture_input_filepaths: list of file paths. + render_input_filepaths: list of file paths. + """ + self._capture_input_filepaths = {} + self._render_input_filepaths = {} + assert len(capture_input_filepaths) == len(render_input_filepaths) + for capture_input_filepath, render_input_filepath in zip( + capture_input_filepaths, render_input_filepaths): + name = self._ExtractFileName(capture_input_filepath) + self._capture_input_filepaths[name] = os.path.abspath( + capture_input_filepath) + self._render_input_filepaths[name] = os.path.abspath( + render_input_filepath) + + @classmethod + def _CreatePathsCollection(cls, filepaths): + """Creates a collection of file paths. + + Given a list of file paths, makes a collection with one item for each file + path. The value is absolute path, the key is the file name without + extenstion. + + Args: + filepaths: list of file paths. + + Returns: + A dict. + """ + filepaths_collection = {} + for filepath in filepaths: + name = cls._ExtractFileName(filepath) + filepaths_collection[name] = os.path.abspath(filepath) + return filepaths_collection + + @classmethod + def _ExtractFileName(cls, filepath): + return os.path.splitext(os.path.split(filepath)[-1])[0] diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py new file mode 100644 index 0000000000..78ca17f589 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py @@ -0,0 +1,203 @@ +# Copyright (c) 2017 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. +"""Unit tests for the simulation module. +""" + +import logging +import os +import shutil +import tempfile +import unittest + +import mock +import pydub + +from . import audioproc_wrapper +from . import eval_scores_factory +from . import evaluation +from . import external_vad +from . import signal_processing +from . import simulation +from . import test_data_generation_factory + + +class TestApmModuleSimulator(unittest.TestCase): + """Unit tests for the ApmModuleSimulator class. + """ + + def setUp(self): + """Create temporary folders and fake audio track.""" + self._output_path = tempfile.mkdtemp() + self._tmp_path = tempfile.mkdtemp() + + silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000) + fake_signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + silence) + self._fake_audio_track_path = os.path.join(self._output_path, + 'fake.wav') + signal_processing.SignalProcessingUtils.SaveWav( + self._fake_audio_track_path, fake_signal) + + def tearDown(self): + """Recursively delete temporary folders.""" + shutil.rmtree(self._output_path) + shutil.rmtree(self._tmp_path) + + def testSimulation(self): + # Instance dependencies to mock and inject. + ap_wrapper = audioproc_wrapper.AudioProcWrapper( + audioproc_wrapper.AudioProcWrapper.DEFAULT_APM_SIMULATOR_BIN_PATH) + evaluator = evaluation.ApmModuleEvaluator() + ap_wrapper.Run = mock.MagicMock(name='Run') + evaluator.Run = mock.MagicMock(name='Run') + + # Instance non-mocked dependencies. + test_data_generator_factory = ( + test_data_generation_factory.TestDataGeneratorFactory( + aechen_ir_database_path='', + noise_tracks_path='', + copy_with_identity=False)) + evaluation_score_factory = eval_scores_factory.EvaluationScoreWorkerFactory( + polqa_tool_bin_path=os.path.join(os.path.dirname(__file__), + 'fake_polqa'), + echo_metric_tool_bin_path=None) + + # Instance simulator. + simulator = simulation.ApmModuleSimulator( + test_data_generator_factory=test_data_generator_factory, + evaluation_score_factory=evaluation_score_factory, + ap_wrapper=ap_wrapper, + evaluator=evaluator, + external_vads={ + 'fake': + external_vad.ExternalVad( + os.path.join(os.path.dirname(__file__), + 'fake_external_vad.py'), 'fake') + }) + + # What to simulate. + config_files = ['apm_configs/default.json'] + input_files = [self._fake_audio_track_path] + test_data_generators = ['identity', 'white_noise'] + eval_scores = ['audio_level_mean', 'polqa'] + + # Run all simulations. + simulator.Run(config_filepaths=config_files, + capture_input_filepaths=input_files, + test_data_generator_names=test_data_generators, + eval_score_names=eval_scores, + output_dir=self._output_path) + + # Check. + # TODO(alessiob): Once the TestDataGenerator classes can be configured by + # the client code (e.g., number of SNR pairs for the white noise test data + # generator), the exact number of calls to ap_wrapper.Run and evaluator.Run + # is known; use that with assertEqual. + min_number_of_simulations = len(config_files) * len(input_files) * len( + test_data_generators) + self.assertGreaterEqual(len(ap_wrapper.Run.call_args_list), + min_number_of_simulations) + self.assertGreaterEqual(len(evaluator.Run.call_args_list), + min_number_of_simulations) + + def testInputSignalCreation(self): + # Instance simulator. + simulator = simulation.ApmModuleSimulator( + test_data_generator_factory=( + test_data_generation_factory.TestDataGeneratorFactory( + aechen_ir_database_path='', + noise_tracks_path='', + copy_with_identity=False)), + evaluation_score_factory=( + eval_scores_factory.EvaluationScoreWorkerFactory( + polqa_tool_bin_path=os.path.join(os.path.dirname(__file__), + 'fake_polqa'), + echo_metric_tool_bin_path=None)), + ap_wrapper=audioproc_wrapper.AudioProcWrapper( + audioproc_wrapper.AudioProcWrapper. + DEFAULT_APM_SIMULATOR_BIN_PATH), + evaluator=evaluation.ApmModuleEvaluator()) + + # Inexistent input files to be silently created. + input_files = [ + os.path.join(self._tmp_path, 'pure_tone-440_1000.wav'), + os.path.join(self._tmp_path, 'pure_tone-1000_500.wav'), + ] + self.assertFalse( + any([os.path.exists(input_file) for input_file in (input_files)])) + + # The input files are created during the simulation. + simulator.Run(config_filepaths=['apm_configs/default.json'], + capture_input_filepaths=input_files, + test_data_generator_names=['identity'], + eval_score_names=['audio_level_peak'], + output_dir=self._output_path) + self.assertTrue( + all([os.path.exists(input_file) for input_file in (input_files)])) + + def testPureToneGenerationWithTotalHarmonicDistorsion(self): + logging.warning = mock.MagicMock(name='warning') + + # Instance simulator. + simulator = simulation.ApmModuleSimulator( + test_data_generator_factory=( + test_data_generation_factory.TestDataGeneratorFactory( + aechen_ir_database_path='', + noise_tracks_path='', + copy_with_identity=False)), + evaluation_score_factory=( + eval_scores_factory.EvaluationScoreWorkerFactory( + polqa_tool_bin_path=os.path.join(os.path.dirname(__file__), + 'fake_polqa'), + echo_metric_tool_bin_path=None)), + ap_wrapper=audioproc_wrapper.AudioProcWrapper( + audioproc_wrapper.AudioProcWrapper. + DEFAULT_APM_SIMULATOR_BIN_PATH), + evaluator=evaluation.ApmModuleEvaluator()) + + # What to simulate. + config_files = ['apm_configs/default.json'] + input_files = [os.path.join(self._tmp_path, 'pure_tone-440_1000.wav')] + eval_scores = ['thd'] + + # Should work. + simulator.Run(config_filepaths=config_files, + capture_input_filepaths=input_files, + test_data_generator_names=['identity'], + eval_score_names=eval_scores, + output_dir=self._output_path) + self.assertFalse(logging.warning.called) + + # Warning expected. + simulator.Run( + config_filepaths=config_files, + capture_input_filepaths=input_files, + test_data_generator_names=['white_noise'], # Not allowed with THD. + eval_score_names=eval_scores, + output_dir=self._output_path) + logging.warning.assert_called_with('the evaluation failed: %s', ( + 'The THD score cannot be used with any test data generator other than ' + '"identity"')) + + # # Init. + # generator = test_data_generation.IdentityTestDataGenerator('tmp') + # input_signal_filepath = os.path.join( + # self._test_data_cache_path, 'pure_tone-440_1000.wav') + + # # Check that the input signal is generated. + # self.assertFalse(os.path.exists(input_signal_filepath)) + # generator.Generate( + # input_signal_filepath=input_signal_filepath, + # test_data_cache_path=self._test_data_cache_path, + # base_output_path=self._base_output_path) + # self.assertTrue(os.path.exists(input_signal_filepath)) + + # # Check input signal properties. + # input_signal = signal_processing.SignalProcessingUtils.LoadWav( + # input_signal_filepath) + # self.assertEqual(1000, len(input_signal)) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc new file mode 100644 index 0000000000..1f24d9d370 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc @@ -0,0 +1,127 @@ +// Copyright (c) 2018 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 +#include +#include +#include + +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "common_audio/include/audio_util.h" +#include "common_audio/wav_file.h" +#include "rtc_base/logging.h" + +ABSL_FLAG(std::string, i, "", "Input wav file"); +ABSL_FLAG(std::string, oc, "", "Config output file"); +ABSL_FLAG(std::string, ol, "", "Levels output file"); +ABSL_FLAG(float, a, 5.f, "Attack (ms)"); +ABSL_FLAG(float, d, 20.f, "Decay (ms)"); +ABSL_FLAG(int, f, 10, "Frame length (ms)"); + +namespace webrtc { +namespace test { +namespace { + +constexpr int kMaxSampleRate = 48000; +constexpr uint8_t kMaxFrameLenMs = 30; +constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000; + +const double kOneDbReduction = DbToRatio(-1.0); + +int main(int argc, char* argv[]) { + absl::ParseCommandLine(argc, argv); + // Check parameters. + if (absl::GetFlag(FLAGS_f) < 1 || absl::GetFlag(FLAGS_f) > kMaxFrameLenMs) { + RTC_LOG(LS_ERROR) << "Invalid frame length (min: 1, max: " << kMaxFrameLenMs + << ")"; + return 1; + } + if (absl::GetFlag(FLAGS_a) < 0 || absl::GetFlag(FLAGS_d) < 0) { + RTC_LOG(LS_ERROR) << "Attack and decay must be non-negative"; + return 1; + } + + // Open wav input file and check properties. + const std::string input_file = absl::GetFlag(FLAGS_i); + const std::string config_output_file = absl::GetFlag(FLAGS_oc); + const std::string levels_output_file = absl::GetFlag(FLAGS_ol); + WavReader wav_reader(input_file); + if (wav_reader.num_channels() != 1) { + RTC_LOG(LS_ERROR) << "Only mono wav files supported"; + return 1; + } + if (wav_reader.sample_rate() > kMaxSampleRate) { + RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate + << ")"; + return 1; + } + + // Map from milliseconds to samples. + const size_t audio_frame_length = rtc::CheckedDivExact( + absl::GetFlag(FLAGS_f) * wav_reader.sample_rate(), 1000); + auto time_const = [](double c) { + return std::pow(kOneDbReduction, absl::GetFlag(FLAGS_f) / c); + }; + const float attack = + absl::GetFlag(FLAGS_a) == 0.0 ? 0.0 : time_const(absl::GetFlag(FLAGS_a)); + const float decay = + absl::GetFlag(FLAGS_d) == 0.0 ? 0.0 : time_const(absl::GetFlag(FLAGS_d)); + + // Write config to file. + std::ofstream out_config(config_output_file); + out_config << "{" + "'frame_len_ms': " + << absl::GetFlag(FLAGS_f) + << ", " + "'attack_ms': " + << absl::GetFlag(FLAGS_a) + << ", " + "'decay_ms': " + << absl::GetFlag(FLAGS_d) << "}\n"; + out_config.close(); + + // Measure level frame-by-frame. + std::ofstream out_levels(levels_output_file, std::ofstream::binary); + std::array samples; + float level_prev = 0.f; + while (true) { + // Process frame. + const auto read_samples = + wav_reader.ReadSamples(audio_frame_length, samples.data()); + if (read_samples < audio_frame_length) + break; // EOF. + + // Frame peak level. + std::transform(samples.begin(), samples.begin() + audio_frame_length, + samples.begin(), [](int16_t s) { return std::abs(s); }); + const int16_t peak_level = *std::max_element( + samples.cbegin(), samples.cbegin() + audio_frame_length); + const float level_curr = static_cast(peak_level) / 32768.f; + + // Temporal smoothing. + auto smooth = [&level_prev, &level_curr](float c) { + return (1.0 - c) * level_curr + c * level_prev; + }; + level_prev = smooth(level_curr > level_prev ? attack : decay); + + // Write output. + out_levels.write(reinterpret_cast(&level_prev), sizeof(float)); + } + out_levels.close(); + + return 0; +} + +} // namespace +} // namespace test +} // namespace webrtc + +int main(int argc, char* argv[]) { + return webrtc::test::main(argc, argv); +} diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py new file mode 100644 index 0000000000..7e86faccec --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py @@ -0,0 +1,526 @@ +# Copyright (c) 2017 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. +"""Test data generators producing signals pairs intended to be used to +test the APM module. Each pair consists of a noisy input and a reference signal. +The former is used as APM input and it is generated by adding noise to a +clean audio track. The reference is the expected APM output. + +Throughout this file, the following naming convention is used: + - input signal: the clean signal (e.g., speech), + - noise signal: the noise to be summed up to the input signal (e.g., white + noise, Gaussian noise), + - noisy signal: input + noise. +The noise signal may or may not be a function of the clean signal. For +instance, white noise is independently generated, whereas reverberation is +obtained by convolving the input signal with an impulse response. +""" + +import logging +import os +import shutil +import sys + +try: + import scipy.io +except ImportError: + logging.critical('Cannot import the third-party Python package scipy') + sys.exit(1) + +from . import data_access +from . import exceptions +from . import signal_processing + + +class TestDataGenerator(object): + """Abstract class responsible for the generation of noisy signals. + + Given a clean signal, it generates two streams named noisy signal and + reference. The former is the clean signal deteriorated by the noise source, + the latter goes through the same deterioration process, but more "gently". + Noisy signal and reference are produced so that the reference is the signal + expected at the output of the APM module when the latter is fed with the noisy + signal. + + An test data generator generates one or more pairs. + """ + + NAME = None + REGISTERED_CLASSES = {} + + def __init__(self, output_directory_prefix): + self._output_directory_prefix = output_directory_prefix + # Init dictionaries with one entry for each test data generator + # configuration (e.g., different SNRs). + # Noisy audio track files (stored separately in a cache folder). + self._noisy_signal_filepaths = None + # Path to be used for the APM simulation output files. + self._apm_output_paths = None + # Reference audio track files (stored separately in a cache folder). + self._reference_signal_filepaths = None + self.Clear() + + @classmethod + def RegisterClass(cls, class_to_register): + """Registers a TestDataGenerator implementation. + + Decorator to automatically register the classes that extend + TestDataGenerator. + Example usage: + + @TestDataGenerator.RegisterClass + class IdentityGenerator(TestDataGenerator): + pass + """ + cls.REGISTERED_CLASSES[class_to_register.NAME] = class_to_register + return class_to_register + + @property + def config_names(self): + return self._noisy_signal_filepaths.keys() + + @property + def noisy_signal_filepaths(self): + return self._noisy_signal_filepaths + + @property + def apm_output_paths(self): + return self._apm_output_paths + + @property + def reference_signal_filepaths(self): + return self._reference_signal_filepaths + + def Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + """Generates a set of noisy input and reference audiotrack file pairs. + + This method initializes an empty set of pairs and calls the _Generate() + method implemented in a concrete class. + + Args: + input_signal_filepath: path to the clean input audio track file. + test_data_cache_path: path to the cache of the generated audio track + files. + base_output_path: base path where output is written. + """ + self.Clear() + self._Generate(input_signal_filepath, test_data_cache_path, + base_output_path) + + def Clear(self): + """Clears the generated output path dictionaries. + """ + self._noisy_signal_filepaths = {} + self._apm_output_paths = {} + self._reference_signal_filepaths = {} + + def _Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + """Abstract method to be implemented in each concrete class. + """ + raise NotImplementedError() + + def _AddNoiseSnrPairs(self, base_output_path, noisy_mix_filepaths, + snr_value_pairs): + """Adds noisy-reference signal pairs. + + Args: + base_output_path: noisy tracks base output path. + noisy_mix_filepaths: nested dictionary of noisy signal paths organized + by noisy track name and SNR level. + snr_value_pairs: list of SNR pairs. + """ + for noise_track_name in noisy_mix_filepaths: + for snr_noisy, snr_refence in snr_value_pairs: + config_name = '{0}_{1:d}_{2:d}_SNR'.format( + noise_track_name, snr_noisy, snr_refence) + output_path = self._MakeDir(base_output_path, config_name) + self._AddNoiseReferenceFilesPair( + config_name=config_name, + noisy_signal_filepath=noisy_mix_filepaths[noise_track_name] + [snr_noisy], + reference_signal_filepath=noisy_mix_filepaths[ + noise_track_name][snr_refence], + output_path=output_path) + + def _AddNoiseReferenceFilesPair(self, config_name, noisy_signal_filepath, + reference_signal_filepath, output_path): + """Adds one noisy-reference signal pair. + + Args: + config_name: name of the APM configuration. + noisy_signal_filepath: path to noisy audio track file. + reference_signal_filepath: path to reference audio track file. + output_path: APM output path. + """ + assert config_name not in self._noisy_signal_filepaths + self._noisy_signal_filepaths[config_name] = os.path.abspath( + noisy_signal_filepath) + self._apm_output_paths[config_name] = os.path.abspath(output_path) + self._reference_signal_filepaths[config_name] = os.path.abspath( + reference_signal_filepath) + + def _MakeDir(self, base_output_path, test_data_generator_config_name): + output_path = os.path.join( + base_output_path, + self._output_directory_prefix + test_data_generator_config_name) + data_access.MakeDirectory(output_path) + return output_path + + +@TestDataGenerator.RegisterClass +class IdentityTestDataGenerator(TestDataGenerator): + """Generator that adds no noise. + + Both the noisy and the reference signals are the input signal. + """ + + NAME = 'identity' + + def __init__(self, output_directory_prefix, copy_with_identity): + TestDataGenerator.__init__(self, output_directory_prefix) + self._copy_with_identity = copy_with_identity + + @property + def copy_with_identity(self): + return self._copy_with_identity + + def _Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + config_name = 'default' + output_path = self._MakeDir(base_output_path, config_name) + + if self._copy_with_identity: + input_signal_filepath_new = os.path.join( + test_data_cache_path, + os.path.split(input_signal_filepath)[1]) + logging.info('copying ' + input_signal_filepath + ' to ' + + (input_signal_filepath_new)) + shutil.copy(input_signal_filepath, input_signal_filepath_new) + input_signal_filepath = input_signal_filepath_new + + self._AddNoiseReferenceFilesPair( + config_name=config_name, + noisy_signal_filepath=input_signal_filepath, + reference_signal_filepath=input_signal_filepath, + output_path=output_path) + + +@TestDataGenerator.RegisterClass +class WhiteNoiseTestDataGenerator(TestDataGenerator): + """Generator that adds white noise. + """ + + NAME = 'white_noise' + + # Each pair indicates the clean vs. noisy and reference vs. noisy SNRs. + # The reference (second value of each pair) always has a lower amount of noise + # - i.e., the SNR is 10 dB higher. + _SNR_VALUE_PAIRS = [ + [20, 30], # Smallest noise. + [10, 20], + [5, 15], + [0, 10], # Largest noise. + ] + + _NOISY_SIGNAL_FILENAME_TEMPLATE = 'noise_{0:d}_SNR.wav' + + def __init__(self, output_directory_prefix): + TestDataGenerator.__init__(self, output_directory_prefix) + + def _Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + # Load the input signal. + input_signal = signal_processing.SignalProcessingUtils.LoadWav( + input_signal_filepath) + + # Create the noise track. + noise_signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise( + input_signal) + + # Create the noisy mixes (once for each unique SNR value). + noisy_mix_filepaths = {} + snr_values = set( + [snr for pair in self._SNR_VALUE_PAIRS for snr in pair]) + for snr in snr_values: + noisy_signal_filepath = os.path.join( + test_data_cache_path, + self._NOISY_SIGNAL_FILENAME_TEMPLATE.format(snr)) + + # Create and save if not done. + if not os.path.exists(noisy_signal_filepath): + # Create noisy signal. + noisy_signal = signal_processing.SignalProcessingUtils.MixSignals( + input_signal, noise_signal, snr) + + # Save. + signal_processing.SignalProcessingUtils.SaveWav( + noisy_signal_filepath, noisy_signal) + + # Add file to the collection of mixes. + noisy_mix_filepaths[snr] = noisy_signal_filepath + + # Add all the noisy-reference signal pairs. + for snr_noisy, snr_refence in self._SNR_VALUE_PAIRS: + config_name = '{0:d}_{1:d}_SNR'.format(snr_noisy, snr_refence) + output_path = self._MakeDir(base_output_path, config_name) + self._AddNoiseReferenceFilesPair( + config_name=config_name, + noisy_signal_filepath=noisy_mix_filepaths[snr_noisy], + reference_signal_filepath=noisy_mix_filepaths[snr_refence], + output_path=output_path) + + +# TODO(alessiob): remove comment when class implemented. +# @TestDataGenerator.RegisterClass +class NarrowBandNoiseTestDataGenerator(TestDataGenerator): + """Generator that adds narrow-band noise. + """ + + NAME = 'narrow_band_noise' + + def __init__(self, output_directory_prefix): + TestDataGenerator.__init__(self, output_directory_prefix) + + def _Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + # TODO(alessiob): implement. + pass + + +@TestDataGenerator.RegisterClass +class AdditiveNoiseTestDataGenerator(TestDataGenerator): + """Generator that adds noise loops. + + This generator uses all the wav files in a given path (default: noise_tracks/) + and mixes them to the clean speech with different target SNRs (hard-coded). + """ + + NAME = 'additive_noise' + _NOISY_SIGNAL_FILENAME_TEMPLATE = '{0}_{1:d}_SNR.wav' + + DEFAULT_NOISE_TRACKS_PATH = os.path.join(os.path.dirname(__file__), + os.pardir, 'noise_tracks') + + # TODO(alessiob): Make the list of SNR pairs customizable. + # Each pair indicates the clean vs. noisy and reference vs. noisy SNRs. + # The reference (second value of each pair) always has a lower amount of noise + # - i.e., the SNR is 10 dB higher. + _SNR_VALUE_PAIRS = [ + [20, 30], # Smallest noise. + [10, 20], + [5, 15], + [0, 10], # Largest noise. + ] + + def __init__(self, output_directory_prefix, noise_tracks_path): + TestDataGenerator.__init__(self, output_directory_prefix) + self._noise_tracks_path = noise_tracks_path + self._noise_tracks_file_names = [ + n for n in os.listdir(self._noise_tracks_path) + if n.lower().endswith('.wav') + ] + if len(self._noise_tracks_file_names) == 0: + raise exceptions.InitializationException( + 'No wav files found in the noise tracks path %s' % + (self._noise_tracks_path)) + + def _Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + """Generates test data pairs using environmental noise. + + For each noise track and pair of SNR values, the following two audio tracks + are created: the noisy signal and the reference signal. The former is + obtained by mixing the (clean) input signal to the corresponding noise + track enforcing the target SNR. + """ + # Init. + snr_values = set( + [snr for pair in self._SNR_VALUE_PAIRS for snr in pair]) + + # Load the input signal. + input_signal = signal_processing.SignalProcessingUtils.LoadWav( + input_signal_filepath) + + noisy_mix_filepaths = {} + for noise_track_filename in self._noise_tracks_file_names: + # Load the noise track. + noise_track_name, _ = os.path.splitext(noise_track_filename) + noise_track_filepath = os.path.join(self._noise_tracks_path, + noise_track_filename) + if not os.path.exists(noise_track_filepath): + logging.error('cannot find the <%s> noise track', + noise_track_filename) + raise exceptions.FileNotFoundError() + + noise_signal = signal_processing.SignalProcessingUtils.LoadWav( + noise_track_filepath) + + # Create the noisy mixes (once for each unique SNR value). + noisy_mix_filepaths[noise_track_name] = {} + for snr in snr_values: + noisy_signal_filepath = os.path.join( + test_data_cache_path, + self._NOISY_SIGNAL_FILENAME_TEMPLATE.format( + noise_track_name, snr)) + + # Create and save if not done. + if not os.path.exists(noisy_signal_filepath): + # Create noisy signal. + noisy_signal = signal_processing.SignalProcessingUtils.MixSignals( + input_signal, + noise_signal, + snr, + pad_noise=signal_processing.SignalProcessingUtils. + MixPadding.LOOP) + + # Save. + signal_processing.SignalProcessingUtils.SaveWav( + noisy_signal_filepath, noisy_signal) + + # Add file to the collection of mixes. + noisy_mix_filepaths[noise_track_name][ + snr] = noisy_signal_filepath + + # Add all the noise-SNR pairs. + self._AddNoiseSnrPairs(base_output_path, noisy_mix_filepaths, + self._SNR_VALUE_PAIRS) + + +@TestDataGenerator.RegisterClass +class ReverberationTestDataGenerator(TestDataGenerator): + """Generator that adds reverberation noise. + + TODO(alessiob): Make this class more generic since the impulse response can be + anything (not just reverberation); call it e.g., + ConvolutionalNoiseTestDataGenerator. + """ + + NAME = 'reverberation' + + _IMPULSE_RESPONSES = { + 'lecture': 'air_binaural_lecture_0_0_1.mat', # Long echo. + 'booth': 'air_binaural_booth_0_0_1.mat', # Short echo. + } + _MAX_IMPULSE_RESPONSE_LENGTH = None + + # Each pair indicates the clean vs. noisy and reference vs. noisy SNRs. + # The reference (second value of each pair) always has a lower amount of noise + # - i.e., the SNR is 5 dB higher. + _SNR_VALUE_PAIRS = [ + [3, 8], # Smallest noise. + [-3, 2], # Largest noise. + ] + + _NOISE_TRACK_FILENAME_TEMPLATE = '{0}.wav' + _NOISY_SIGNAL_FILENAME_TEMPLATE = '{0}_{1:d}_SNR.wav' + + def __init__(self, output_directory_prefix, aechen_ir_database_path): + TestDataGenerator.__init__(self, output_directory_prefix) + self._aechen_ir_database_path = aechen_ir_database_path + + def _Generate(self, input_signal_filepath, test_data_cache_path, + base_output_path): + """Generates test data pairs using reverberation noise. + + For each impulse response, one noise track is created. For each impulse + response and pair of SNR values, the following 2 audio tracks are + created: the noisy signal and the reference signal. The former is + obtained by mixing the (clean) input signal to the corresponding noise + track enforcing the target SNR. + """ + # Init. + snr_values = set( + [snr for pair in self._SNR_VALUE_PAIRS for snr in pair]) + + # Load the input signal. + input_signal = signal_processing.SignalProcessingUtils.LoadWav( + input_signal_filepath) + + noisy_mix_filepaths = {} + for impulse_response_name in self._IMPULSE_RESPONSES: + noise_track_filename = self._NOISE_TRACK_FILENAME_TEMPLATE.format( + impulse_response_name) + noise_track_filepath = os.path.join(test_data_cache_path, + noise_track_filename) + noise_signal = None + try: + # Load noise track. + noise_signal = signal_processing.SignalProcessingUtils.LoadWav( + noise_track_filepath) + except exceptions.FileNotFoundError: + # Generate noise track by applying the impulse response. + impulse_response_filepath = os.path.join( + self._aechen_ir_database_path, + self._IMPULSE_RESPONSES[impulse_response_name]) + noise_signal = self._GenerateNoiseTrack( + noise_track_filepath, input_signal, + impulse_response_filepath) + assert noise_signal is not None + + # Create the noisy mixes (once for each unique SNR value). + noisy_mix_filepaths[impulse_response_name] = {} + for snr in snr_values: + noisy_signal_filepath = os.path.join( + test_data_cache_path, + self._NOISY_SIGNAL_FILENAME_TEMPLATE.format( + impulse_response_name, snr)) + + # Create and save if not done. + if not os.path.exists(noisy_signal_filepath): + # Create noisy signal. + noisy_signal = signal_processing.SignalProcessingUtils.MixSignals( + input_signal, noise_signal, snr) + + # Save. + signal_processing.SignalProcessingUtils.SaveWav( + noisy_signal_filepath, noisy_signal) + + # Add file to the collection of mixes. + noisy_mix_filepaths[impulse_response_name][ + snr] = noisy_signal_filepath + + # Add all the noise-SNR pairs. + self._AddNoiseSnrPairs(base_output_path, noisy_mix_filepaths, + self._SNR_VALUE_PAIRS) + + def _GenerateNoiseTrack(self, noise_track_filepath, input_signal, + impulse_response_filepath): + """Generates noise track. + + Generate a signal by convolving input_signal with the impulse response in + impulse_response_filepath; then save to noise_track_filepath. + + Args: + noise_track_filepath: output file path for the noise track. + input_signal: (clean) input signal samples. + impulse_response_filepath: impulse response file path. + + Returns: + AudioSegment instance. + """ + # Load impulse response. + data = scipy.io.loadmat(impulse_response_filepath) + impulse_response = data['h_air'].flatten() + if self._MAX_IMPULSE_RESPONSE_LENGTH is not None: + logging.info('truncating impulse response from %d to %d samples', + len(impulse_response), + self._MAX_IMPULSE_RESPONSE_LENGTH) + impulse_response = impulse_response[:self. + _MAX_IMPULSE_RESPONSE_LENGTH] + + # Apply impulse response. + processed_signal = ( + signal_processing.SignalProcessingUtils.ApplyImpulseResponse( + input_signal, impulse_response)) + + # Save. + signal_processing.SignalProcessingUtils.SaveWav( + noise_track_filepath, processed_signal) + + return processed_signal diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py new file mode 100644 index 0000000000..948888e775 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py @@ -0,0 +1,71 @@ +# Copyright (c) 2017 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. +"""TestDataGenerator factory class. +""" + +import logging + +from . import exceptions +from . import test_data_generation + + +class TestDataGeneratorFactory(object): + """Factory class used to create test data generators. + + Usage: Create a factory passing parameters to the ctor with which the + generators will be produced. + """ + + def __init__(self, aechen_ir_database_path, noise_tracks_path, + copy_with_identity): + """Ctor. + + Args: + aechen_ir_database_path: Path to the Aechen Impulse Response database. + noise_tracks_path: Path to the noise tracks to add. + copy_with_identity: Flag indicating whether the identity generator has to + make copies of the clean speech input files. + """ + self._output_directory_prefix = None + self._aechen_ir_database_path = aechen_ir_database_path + self._noise_tracks_path = noise_tracks_path + self._copy_with_identity = copy_with_identity + + def SetOutputDirectoryPrefix(self, prefix): + self._output_directory_prefix = prefix + + def GetInstance(self, test_data_generators_class): + """Creates an TestDataGenerator instance given a class object. + + Args: + test_data_generators_class: TestDataGenerator class object (not an + instance). + + Returns: + TestDataGenerator instance. + """ + if self._output_directory_prefix is None: + raise exceptions.InitializationException( + 'The output directory prefix for test data generators is not set' + ) + logging.debug('factory producing %s', test_data_generators_class) + + if test_data_generators_class == ( + test_data_generation.IdentityTestDataGenerator): + return test_data_generation.IdentityTestDataGenerator( + self._output_directory_prefix, self._copy_with_identity) + elif test_data_generators_class == ( + test_data_generation.ReverberationTestDataGenerator): + return test_data_generation.ReverberationTestDataGenerator( + self._output_directory_prefix, self._aechen_ir_database_path) + elif test_data_generators_class == ( + test_data_generation.AdditiveNoiseTestDataGenerator): + return test_data_generation.AdditiveNoiseTestDataGenerator( + self._output_directory_prefix, self._noise_tracks_path) + else: + return test_data_generators_class(self._output_directory_prefix) diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py new file mode 100644 index 0000000000..f75098ae2c --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py @@ -0,0 +1,207 @@ +# Copyright (c) 2017 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. +"""Unit tests for the test_data_generation module. +""" + +import os +import shutil +import tempfile +import unittest + +import numpy as np +import scipy.io + +from . import test_data_generation +from . import test_data_generation_factory +from . import signal_processing + + +class TestTestDataGenerators(unittest.TestCase): + """Unit tests for the test_data_generation module. + """ + + def setUp(self): + """Create temporary folders.""" + self._base_output_path = tempfile.mkdtemp() + self._test_data_cache_path = tempfile.mkdtemp() + self._fake_air_db_path = tempfile.mkdtemp() + + # Fake AIR DB impulse responses. + # TODO(alessiob): ReverberationTestDataGenerator will change to allow custom + # impulse responses. When changed, the coupling below between + # impulse_response_mat_file_names and + # ReverberationTestDataGenerator._IMPULSE_RESPONSES can be removed. + impulse_response_mat_file_names = [ + 'air_binaural_lecture_0_0_1.mat', + 'air_binaural_booth_0_0_1.mat', + ] + for impulse_response_mat_file_name in impulse_response_mat_file_names: + data = {'h_air': np.random.rand(1, 1000).astype(' +#include +#include + +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "common_audio/wav_file.h" +#include "rtc_base/logging.h" + +ABSL_FLAG(std::string, i, "", "Input wav file"); +ABSL_FLAG(std::string, o, "", "VAD output file"); + +namespace webrtc { +namespace test { +namespace { + +// The allowed values are 10, 20 or 30 ms. +constexpr uint8_t kAudioFrameLengthMilliseconds = 30; +constexpr int kMaxSampleRate = 48000; +constexpr size_t kMaxFrameLen = + kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000; + +constexpr uint8_t kBitmaskBuffSize = 8; + +int main(int argc, char* argv[]) { + absl::ParseCommandLine(argc, argv); + const std::string input_file = absl::GetFlag(FLAGS_i); + const std::string output_file = absl::GetFlag(FLAGS_o); + // Open wav input file and check properties. + WavReader wav_reader(input_file); + if (wav_reader.num_channels() != 1) { + RTC_LOG(LS_ERROR) << "Only mono wav files supported"; + return 1; + } + if (wav_reader.sample_rate() > kMaxSampleRate) { + RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate + << ")"; + return 1; + } + const size_t audio_frame_length = rtc::CheckedDivExact( + kAudioFrameLengthMilliseconds * wav_reader.sample_rate(), 1000); + if (audio_frame_length > kMaxFrameLen) { + RTC_LOG(LS_ERROR) << "The frame size and/or the sample rate are too large."; + return 1; + } + + // Create output file and write header. + std::ofstream out_file(output_file, std::ofstream::binary); + const char audio_frame_length_ms = kAudioFrameLengthMilliseconds; + out_file.write(&audio_frame_length_ms, 1); // Header. + + // Run VAD and write decisions. + std::unique_ptr vad = CreateVad(Vad::Aggressiveness::kVadNormal); + std::array samples; + char buff = 0; // Buffer to write one bit per frame. + uint8_t next = 0; // Points to the next bit to write in `buff`. + while (true) { + // Process frame. + const auto read_samples = + wav_reader.ReadSamples(audio_frame_length, samples.data()); + if (read_samples < audio_frame_length) + break; + const auto is_speech = vad->VoiceActivity( + samples.data(), audio_frame_length, wav_reader.sample_rate()); + + // Write output. + buff = is_speech ? buff | (1 << next) : buff & ~(1 << next); + if (++next == kBitmaskBuffSize) { + out_file.write(&buff, 1); // Flush. + buff = 0; // Reset. + next = 0; + } + } + + // Finalize. + char extra_bits = 0; + if (next > 0) { + extra_bits = kBitmaskBuffSize - next; + out_file.write(&buff, 1); // Flush. + } + out_file.write(&extra_bits, 1); + out_file.close(); + + return 0; +} + +} // namespace +} // namespace test +} // namespace webrtc + +int main(int argc, char* argv[]) { + return webrtc::test::main(argc, argv); +} -- cgit v1.2.3