Fix the stereo support in IntelligibilityEnhancer

webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-//
-//  Command line tool for speech intelligibility enhancement. Provides for
-//  running and testing intelligibility_enhancer as an independent process.
-//  Use --help for options.
-//
-
-#include <sys/stat.h>
-
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 
 using std::complex;
 
 namespace webrtc {
 namespace {
 
 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
 DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
 DEFINE_string(out_file, "proc_enhanced.wav", "Enhanced output file.");
 
 // void function for gtest
 void void_main(int argc, char* argv[]) {
   google::SetUsageMessage(
       "\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
   google::ParseCommandLineFlags(&argc, &argv, true);
 
-  // Load settings and wav input.
-  struct stat in_stat, noise_stat;
-  ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
-      << "Empty speech file.";
-  ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
-      << "Empty noise file.";
-
-  const size_t samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
-
   WavReader in_file(FLAGS_clear_file);
-  std::vector<float> in_fpcm(samples);
-  in_file.ReadSamples(samples, &in_fpcm[0]);
-  FloatS16ToFloat(&in_fpcm[0], samples, &in_fpcm[0]);
-
   WavReader noise_file(FLAGS_noise_file);
-  std::vector<float> noise_fpcm(samples);
-  noise_file.ReadSamples(samples, &noise_fpcm[0]);
-  FloatS16ToFloat(&noise_fpcm[0], samples, &noise_fpcm[0]);
-
-  // Run intelligibility enhancement.
+  WavWriter out_file(FLAGS_out_file,
+                     in_file.sample_rate(),
+                     in_file.num_channels());
   IntelligibilityEnhancer enh(in_file.sample_rate(), in_file.num_channels());
   rtc::CriticalSection crit;
   NoiseSuppressionImpl ns(&crit);
   ns.Initialize(noise_file.num_channels(), noise_file.sample_rate());
   ns.Enable(true);
-
-  // Mirror real time APM chunk size. Duplicates chunk_length_ in
-  // IntelligibilityEnhancer.
-  size_t fragment_size = in_file.sample_rate() / 100;
-  AudioBuffer capture_audio(fragment_size, noise_file.num_channels(),
-                            fragment_size, noise_file.num_channels(),
-                            fragment_size);
-  StreamConfig stream_config(in_file.sample_rate(), noise_file.num_channels());
-
-  // Slice the input into smaller chunks, as the APM would do, and feed them
-  // through the enhancer.
-  float* clear_cursor = &in_fpcm[0];
-  float* noise_cursor = &noise_fpcm[0];
-
-  for (size_t i = 0; i < samples; i += fragment_size) {
-    capture_audio.CopyFrom(&noise_cursor, stream_config);
+  const size_t in_samples = noise_file.sample_rate() / 100;
+  const size_t noise_samples = noise_file.sample_rate() / 100;

[hlundin-webrtc, 2016/02/24 10:17:03]

What if the sample rates differ? Is that allowed and handled?

[aluebs-webrtc, 2016/02/25 00:18:37]

Yes it is, since the sample rate of the noise only matters to the NS and the one
of the input only matters to the IE.

[hlundin-webrtc, 2016/02/26 21:58:57]

Acknowledged.

+  std::vector<float> in(in_samples * in_file.num_channels());
+  std::vector<float> noise(noise_samples * noise_file.num_channels());
+  ChannelBuffer<float> in_buf(in_samples, in_file.num_channels());
+  ChannelBuffer<float> noise_buf(noise_samples, noise_file.num_channels());
+  AudioBuffer capture_audio(noise_samples, noise_file.num_channels(),
+                            noise_samples, noise_file.num_channels(),
+                            noise_samples);
+  StreamConfig stream_config(noise_file.sample_rate(),
+                             noise_file.num_channels());
+  while (in_file.ReadSamples(in.size(), &in[0]) == in.size() &&

[hlundin-webrtc, 2016/02/24 10:17:03]

in.data()
several places below

[aluebs-webrtc, 2016/02/25 00:18:37]

Done.

+         noise_file.ReadSamples(noise.size(), &noise[0]) == noise.size()) {

[hlundin-webrtc, 2016/02/24 10:17:03]

noise.data()
several places below

[aluebs-webrtc, 2016/02/25 00:18:37]

Done.

+    FloatS16ToFloat(&in[0], in.size(), &in[0]);
+    FloatS16ToFloat(&noise[0], noise.size(), &noise[0]);

[turaj, 2016/02/24 16:00:17]

NOT RELATED TO THIS CL--Is is guaranteed that in practice near end and far end
signals to APM are in the same range? I guess no other module requires this. 

[aluebs-webrtc, 2016/02/25 00:18:37]

That is a good question. Testing with audioproc they are and I want to hope this
holds for all platforms. It should, since in an aecdump the render and capture
signals are in the same range, right?

+    Deinterleave(&in[0],
+                 in_buf.num_frames(),
+                 in_buf.num_channels(),
+                 in_buf.channels());
+    Deinterleave(&noise[0],
+                 noise_buf.num_frames(),
+                 noise_buf.num_channels(),
+                 noise_buf.channels());
+    capture_audio.CopyFrom(noise_buf.channels(), stream_config);
     ns.AnalyzeCaptureAudio(&capture_audio);
     ns.ProcessCaptureAudio(&capture_audio);
     enh.SetCaptureNoiseEstimate(ns.NoiseEstimate());
-    enh.ProcessRenderAudio(&clear_cursor, in_file.sample_rate(),
+    enh.ProcessRenderAudio(in_buf.channels(),
+                           in_file.sample_rate(),
                            in_file.num_channels());
-    clear_cursor += fragment_size;
-    noise_cursor += fragment_size;
+    Interleave(in_buf.channels(),
+               in_buf.num_frames(),
+               in_buf.num_channels(),
+               &in[0]);
+    FloatToFloatS16(&in[0], in.size(), &in[0]);
+    out_file.WriteSamples(&in[0], in.size());
   }
-
-  FloatToFloatS16(&in_fpcm[0], samples, &in_fpcm[0]);
-
-  WavWriter out_file(FLAGS_out_file,
-                     in_file.sample_rate(),
-                     in_file.num_channels());
-  out_file.WriteSamples(&in_fpcm[0], samples);
 }
 
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
   return 0;
 }