Remove unused audio_e2e_harness.cc and infrastructure.
BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2295283004 Cr-Commit-Position: refs/heads/master@{#14165}
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@ -219,28 +219,6 @@ if (rtc_include_tests) {
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]
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}
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rtc_executable("audio_e2e_harness") {
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testonly = true
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sources = [
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"e2e_quality/audio/audio_e2e_harness.cc",
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]
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if (is_clang) {
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# Suppress warnings from the Chromium Clang plugin.
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# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../system_wrappers:system_wrappers_default",
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"../test:channel_transport",
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"../voice_engine",
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"//build/win:default_exe_manifest",
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"//testing/gtest",
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"//third_party/gflags",
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]
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}
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if (is_android || is_ios) {
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tools_unittests_resources = [ "//resources/foreman_cif.yuv" ]
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}
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@ -6,6 +6,5 @@ include_rules = [
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"+webrtc/modules/congestion_controller",
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"+webrtc/modules/rtp_rtcp",
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"+webrtc/system_wrappers",
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"+webrtc/voice_engine",
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]
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@ -1,27 +0,0 @@
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The tools here run an end-to-end audio quality test on Linux using PulseAudio.
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INSTALLATION
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The test depends on PulseAudio virtual devices (null sinks). Without additional
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arguments, run_audio_test.py expects a pair of sinks named "capture" and
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"render". To create these devices at machine startup, place the provided
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default.pa file in ~/.pulse. Alternately, the "pacmd" commands therein can be
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run on the command-line to create the devices.
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Similarly, place the provided daemon.conf file in ~/.pulse to use high quality
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resampling in PulseAudio. This will reduce the resampling impact on the outcome
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of the test.
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Build all WebRTC targets as usual (or just the audio_e2e_harness target) to
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generate the VoiceEngine harness.
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USAGE
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Run run_audio_test.py to start. The script has reasonable defaults and will
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use the expected location of audio_e2e_harness. Some settings will usually
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be provided by the user, particularly the comparison tool command-line and
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regular expression to extract the quality metric.
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An example command-line, run from trunk/
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webrtc/tools/e2e_quality/audio/run_audio_test.py \
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--input=data/voice_engine/audio_short16.pcm --output=e2e_audio_out.pcm \
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--codec=L16 --compare="comparison-tool" --regexp="(\d\.\d{3})"
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@ -1,109 +0,0 @@
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Sets up a simple VoiceEngine loopback call with the default audio devices
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// and runs forever. Some parameters can be configured through command-line
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// flags.
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#include <memory>
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#include "gflags/gflags.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/test/channel_transport/channel_transport.h"
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_codec.h"
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#include "webrtc/voice_engine/include/voe_hardware.h"
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#include "webrtc/voice_engine/include/voe_network.h"
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DEFINE_string(render, "render", "render device name");
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DEFINE_string(codec, "ISAC", "codec name");
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DEFINE_int32(rate, 16000, "codec sample rate in Hz");
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namespace webrtc {
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namespace test {
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void RunHarness() {
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VoiceEngine* voe = VoiceEngine::Create();
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ASSERT_TRUE(voe != NULL);
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VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe);
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ASSERT_TRUE(audio != NULL);
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VoEBase* base = VoEBase::GetInterface(voe);
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ASSERT_TRUE(base != NULL);
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VoECodec* codec = VoECodec::GetInterface(voe);
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ASSERT_TRUE(codec != NULL);
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VoEHardware* hardware = VoEHardware::GetInterface(voe);
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ASSERT_TRUE(hardware != NULL);
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VoENetwork* network = VoENetwork::GetInterface(voe);
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ASSERT_TRUE(network != NULL);
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ASSERT_EQ(0, base->Init());
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int channel = base->CreateChannel();
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ASSERT_NE(-1, channel);
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std::unique_ptr<VoiceChannelTransport> voice_channel_transport(
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new VoiceChannelTransport(network, channel));
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ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
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ASSERT_EQ(0, voice_channel_transport->SetLocalReceiver(1234));
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CodecInst codec_params = {0};
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bool codec_found = false;
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for (int i = 0; i < codec->NumOfCodecs(); i++) {
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ASSERT_EQ(0, codec->GetCodec(i, codec_params));
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if (FLAGS_codec.compare(codec_params.plname) == 0 &&
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FLAGS_rate == codec_params.plfreq) {
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codec_found = true;
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break;
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}
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}
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ASSERT_TRUE(codec_found);
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ASSERT_EQ(0, codec->SetSendCodec(channel, codec_params));
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int num_devices = 0;
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ASSERT_EQ(0, hardware->GetNumOfPlayoutDevices(num_devices));
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char device_name[128] = {0};
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char guid[128] = {0};
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bool device_found = false;
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int device_index;
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for (device_index = 0; device_index < num_devices; device_index++) {
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ASSERT_EQ(0, hardware->GetPlayoutDeviceName(device_index, device_name,
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guid));
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if (FLAGS_render.compare(device_name) == 0) {
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device_found = true;
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break;
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}
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}
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ASSERT_TRUE(device_found);
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ASSERT_EQ(0, hardware->SetPlayoutDevice(device_index));
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// Disable all audio processing.
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ASSERT_EQ(0, audio->SetAgcStatus(false));
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ASSERT_EQ(0, audio->SetEcStatus(false));
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ASSERT_EQ(0, audio->EnableHighPassFilter(false));
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ASSERT_EQ(0, audio->SetNsStatus(false));
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ASSERT_EQ(0, base->StartReceive(channel));
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ASSERT_EQ(0, base->StartPlayout(channel));
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ASSERT_EQ(0, base->StartSend(channel));
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// Run forever...
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while (1) {
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}
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}
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} // namespace test
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} // namespace webrtc
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int main(int argc, char** argv) {
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google::ParseCommandLineFlags(&argc, &argv, true);
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webrtc::test::RunHarness();
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}
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@ -1 +0,0 @@
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resample-method = speex-float-9
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@ -1,6 +0,0 @@
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# Place in ~/.pulse/ to add null sinks for the audio end-to-end quality test.
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.include /etc/pulse/default.pa
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load-module module-null-sink sink_name=render sink_properties=device.description=render format=s16 rate=48000 channels=1
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load-module module-null-sink sink_name=capture sink_properties=device.description=capture format=s16 rate=48000 channels=1
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@ -1 +0,0 @@
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../../../../tools/perf
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@ -1,131 +0,0 @@
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#!/usr/bin/env python
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#
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# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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"""Runs an end-to-end audio quality test on Linux.
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Expects the presence of PulseAudio virtual devices (null sinks). These are
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configured as default devices for a VoiceEngine audio call. A PulseAudio
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utility (pacat) is used to play to and record from the virtual devices.
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The input reference file is then compared to the output file.
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"""
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import optparse
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import os
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import re
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import shlex
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import subprocess
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import sys
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import time
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import perf.perf_utils
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def main(argv):
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parser = optparse.OptionParser()
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usage = 'Usage: %prog [options]'
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parser.set_usage(usage)
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parser.add_option('--input', default='input.pcm', help='input PCM file')
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parser.add_option('--output', default='output.pcm', help='output PCM file')
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parser.add_option('--codec', default='ISAC', help='codec name')
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parser.add_option('--rate', default='16000', help='sample rate in Hz')
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parser.add_option('--channels', default='1', help='number of channels')
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parser.add_option('--play_sink', default='capture',
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help='name of PulseAudio sink to which to play audio')
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parser.add_option('--rec_sink', default='render',
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help='name of PulseAudio sink whose monitor will be recorded')
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parser.add_option('--harness',
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default=os.path.abspath(os.path.dirname(sys.argv[0]) +
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'/../../../out/Debug/audio_e2e_harness'),
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help='path to audio harness executable')
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parser.add_option('--compare',
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help='command-line arguments for comparison tool')
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parser.add_option('--regexp',
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help='regular expression to extract the comparison metric')
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options, _ = parser.parse_args(argv[1:])
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# Get the initial default capture device, to restore later.
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command = ['pacmd', 'list-sources']
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print ' '.join(command)
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proc = subprocess.Popen(command, stdout=subprocess.PIPE)
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output = proc.communicate()[0]
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if proc.returncode != 0:
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return proc.returncode
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default_source = re.search(r'(^ \* index: )([0-9]+$)', output,
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re.MULTILINE).group(2)
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# Set the default capture device to be used by VoiceEngine. We unfortunately
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# need to do this rather than select the devices directly through the harness
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# because monitor sources don't appear in VoiceEngine except as defaults.
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#
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# We pass the render device for VoiceEngine to select because (for unknown
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# reasons) the virtual device is sometimes not used when the default.
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command = ['pacmd', 'set-default-source', options.play_sink + '.monitor']
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print ' '.join(command)
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retcode = subprocess.call(command, stdout=subprocess.PIPE)
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if retcode != 0:
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return retcode
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command = [options.harness, '--render=' + options.rec_sink,
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'--codec=' + options.codec, '--rate=' + options.rate]
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print ' '.join(command)
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voe_proc = subprocess.Popen(command)
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# If recording starts before there is data available, pacat sometimes
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# inexplicably adds a large delay to the start of the file. We wait here in
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# an attempt to prevent that, because VoE often takes some time to startup a
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# call.
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time.sleep(5)
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format_args = ['--format=s16le', '--rate=' + options.rate,
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'--channels=' + options.channels, '--raw']
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command = (['pacat', '-p', '-d', options.play_sink] + format_args +
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[options.input])
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print ' '.join(command)
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play_proc = subprocess.Popen(command)
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command = (['pacat', '-r', '-d', options.rec_sink + '.monitor'] +
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format_args + [options.output])
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print ' '.join(command)
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record_proc = subprocess.Popen(command)
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retcode = play_proc.wait()
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# If these ended early, an exception will be thrown here.
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record_proc.kill()
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voe_proc.kill()
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if retcode != 0:
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return retcode
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# Restore the initial default capture device.
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command = ['pacmd', 'set-default-source', default_source]
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print ' '.join(command)
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retcode = subprocess.call(command, stdout=subprocess.PIPE)
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if retcode != 0:
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return retcode
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if options.compare and options.regexp:
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command = shlex.split(options.compare) + [options.input, options.output]
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print ' '.join(command)
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proc = subprocess.Popen(command, stdout=subprocess.PIPE)
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output = proc.communicate()[0]
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if proc.returncode != 0:
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return proc.returncode
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# The list should only contain one item.
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value = ''.join(re.findall(options.regexp, output))
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perf.perf_utils.PrintPerfResult(graph_name='audio_e2e_score',
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series_name='e2e_score',
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data_point=value,
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units='MOS') # Assuming we run PESQ.
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return 0
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if __name__ == '__main__':
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sys.exit(main(sys.argv))
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@ -176,20 +176,6 @@
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'agc/activity_metric.cc',
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],
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}, # activity_metric
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{
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'target_name': 'audio_e2e_harness',
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'type': 'executable',
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'dependencies': [
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'<(webrtc_root)/test/test.gyp:channel_transport',
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'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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],
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'sources': [
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'e2e_quality/audio/audio_e2e_harness.cc',
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],
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}, # audio_e2e_harness
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{
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'target_name': 'tools_unittests',
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'type': '<(gtest_target_type)',
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