Add a framework for audio end-to-end quality testing.

The quality comparison step is still to be done.

BUG=issue502
TEST=manual

Review URL: https://webrtc-codereview.appspot.com/577005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2220 4adac7df-926f-26a2-2b94-8c16560cd09d

tools/e2e_quality/audio/audio_e2e_harness.cc

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+// Sets up a simple VoiceEngine loopback call with the default audio devices
+// and runs forever. Some parameters can be configured through command-line
+// flags.
+
+#include "gflags/gflags.h"
+#include "gtest/gtest.h"
+
+#include "src/voice_engine/main/interface/voe_audio_processing.h"
+#include "src/voice_engine/main/interface/voe_base.h"
+#include "src/voice_engine/main/interface/voe_codec.h"
+
+DEFINE_string(codec, "ISAC", "codec name");
+DEFINE_int32(rate, 16000, "codec sample rate in Hz");
+
+namespace webrtc {
+namespace {
+
+void RunHarness() {
+  VoiceEngine* voe = VoiceEngine::Create();
+  ASSERT_TRUE(voe != NULL);
+  VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe);
+  ASSERT_TRUE(audio != NULL);
+  VoEBase* base = VoEBase::GetInterface(voe);
+  ASSERT_TRUE(base != NULL);
+  VoECodec* codec = VoECodec::GetInterface(voe);
+  ASSERT_TRUE(codec != NULL);
+
+  ASSERT_EQ(0, base->Init());
+  int channel = base->CreateChannel();
+  ASSERT_NE(-1, channel);
+  ASSERT_EQ(0, base->SetSendDestination(channel, 1234, "127.0.0.1"));
+  ASSERT_EQ(0, base->SetLocalReceiver(channel, 1234));
+
+  CodecInst codec_params = {0};
+  bool codec_found = false;
+  for (int i = 0; i < codec->NumOfCodecs(); i++) {
+    ASSERT_EQ(0, codec->GetCodec(i, codec_params));
+    if (FLAGS_codec.compare(codec_params.plname) == 0 &&
+        FLAGS_rate == codec_params.plfreq) {
+      codec_found = true;
+      break;
+    }
+  }
+  ASSERT_TRUE(codec_found);
+  ASSERT_EQ(0, codec->SetSendCodec(channel, codec_params));
+
+  // Disable all audio processing.
+  ASSERT_EQ(0, audio->SetAgcStatus(false));
+  ASSERT_EQ(0, audio->SetEcStatus(false));
+  ASSERT_EQ(0, audio->EnableHighPassFilter(false));
+  ASSERT_EQ(0, audio->SetNsStatus(false));
+
+  ASSERT_EQ(0, base->StartReceive(channel));
+  ASSERT_EQ(0, base->StartPlayout(channel));
+  ASSERT_EQ(0, base->StartSend(channel));
+
+  // Run forever...
+  while (1);
+}
+
+}  // namespace
+}  // namespace webrtc
+
+int main(int argc, char** argv) {
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::RunHarness();
+}

tools/e2e_quality/audio/default.pa

@@ -0,0 +1,6 @@
+# Place in ~/.pulse/ to add null sinks for the audio end-to-end quality test.
+
+.include /etc/pulse/default.pa
+
+load-module module-null-sink sink_name=render sink_properties=device.description=render format=s16 rate=48000 channels=1
+load-module module-null-sink sink_name=capture sink_properties=device.description=capture format=s16 rate=48000 channels=1

tools/e2e_quality/audio/run_audio_test.py

@@ -0,0 +1,94 @@
+#!/usr/bin/env python
+# Copyright (c) 2012 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.
+
+__author__ = 'andrew@webrtc.org (Andrew MacDonald)'
+
+import optparse
+import os
+import shlex
+import subprocess
+import sys
+import threading
+import time
+
+"""Runs an end-to-end audio quality test on Linux.
+
+Expects the presence of PulseAudio virtual devices (null sinks). These are
+configured as default devices for a VoiceEngine audio call. A PulseAudio
+utility (pacat) is used to play to and record from the virtual devices.
+
+The input reference file is then compared to the output file.
+"""
+
+def popen_and_call(popen_args, call_on_exit):
+  """Executes the arguments, and triggers the callback when finished."""
+
+  def run_in_thread(popen_args, call_on_exit):
+    proc = subprocess.Popen(popen_args)
+    proc.wait()
+    call_on_exit()
+    return
+  thread = threading.Thread(target=run_in_thread,
+                            args=(popen_args, call_on_exit))
+  thread.start()
+  return thread
+
+def main(argv):
+  parser = optparse.OptionParser()
+  usage = 'Usage: %prog [options]'
+  parser.set_usage(usage)
+  parser.add_option('--input', default='input.pcm', help='input PCM file')
+  parser.add_option('--output', default='output.pcm', help='output PCM file')
+  parser.add_option('--codec', default='ISAC', help='codec name')
+  parser.add_option('--rate', default='16000', help='sample rate in Hz')
+  parser.add_option('--channels', default='1', help='number of channels')
+  parser.add_option('--play_sink', default='capture',
+      help='name of PulseAudio sink to which to play audio')
+  parser.add_option('--rec_sink', default='render',
+      help='name of PulseAudio sink whose monitor will be recorded')
+  parser.add_option('--harness',
+      default=os.path.dirname(sys.argv[0]) +
+          '/../../../out/Debug/audio_e2e_harness',
+      help='path to audio harness executable')
+  (options, args) = parser.parse_args(argv[1:])
+
+  # Set default devices to be used by VoiceEngine.
+  subprocess.call(['pacmd', 'set-default-sink', options.rec_sink]);
+  subprocess.call(['pacmd', 'set-default-source',
+      options.play_sink + '.monitor']);
+
+  print 'Start an audio call'
+  print options.harness
+  voe_proc = subprocess.Popen([options.harness,
+      '--codec=' + options.codec, '--rate=' + options.rate]);
+
+  print 'Start recording to ' + options.output
+  format_args = ('-n --format=s16le --rate=' + options.rate + ' --channels=' +
+      options.channels + ' --raw')
+  command = ('pacat -r -d ' + options.rec_sink + '.monitor ' + format_args +
+      ' ' + options.output)
+  record_proc = subprocess.Popen(shlex.split(command))
+  def stop_recording():
+    record_proc.kill()
+
+  print 'Start playing from ' + options.input
+  command = ('pacat -p -d ' + options.play_sink + ' ' + format_args + ' ' +
+      options.input)
+  popen_and_call(shlex.split(command), stop_recording)
+
+  # record_proc will be killed after playout finishes.
+  record_proc.wait()
+
+  print 'Shutdown audio call'
+  voe_proc.kill()
+
+  # TODO(andrew): compare files.
+
+if __name__ == '__main__':
+  sys.exit(main(sys.argv))

tools/e2e_quality/e2e_quality.gyp

@@ -0,0 +1,25 @@
+# Copyright (c) 2012 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.
+
+{
+  'includes': [ '../../src/build/common.gypi'],
+  'targets': [
+    {
+      'target_name': 'audio_e2e_harness',
+      'type': 'executable',
+      'dependencies': [
+        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
+      ],
+      'sources': [
+        'audio/audio_e2e_harness.cc',
+      ],
+    },
+  ],
+}

webrtc.gyp

@@ -21,9 +21,8 @@
         'src/voice_engine/voice_engine.gyp:*',
         'test/metrics.gyp:*',
         'test/test.gyp:*',
+        'tools/e2e_quality/e2e_quality.gyp:*',
       ],
     },
   ],
-  'conditions': [
-  ],  # conditions
 }