Revert of Add aecdump support to audioproc_f.

webrtc/modules/audio_processing/test/audioproc_float.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.
  */
 
 #include <stdio.h>
-#include <iostream>
 #include <sstream>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
-DEFINE_string(dump, "", "Name of the aecdump debug file to read from.");
-DEFINE_string(i, "", "Name of the capture input stream file to read from.");
-DEFINE_string(
-    o,
-    "out.wav",
-    "Name of the output file to write the processed capture stream to.");
-DEFINE_int32(out_channels, 1, "Number of output channels.");
-DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz.");
+DEFINE_string(dump, "", "The name of the debug dump file to read from.");
+DEFINE_string(i, "", "The name of the input file to read from.");
+DEFINE_string(i_rev, "", "The name of the reverse input file to read from.");
+DEFINE_string(o, "out.wav", "Name of the output file to write to.");
+DEFINE_string(o_rev,
+              "out_rev.wav",
+              "Name of the reverse output file to write to.");
+DEFINE_int32(out_channels, 0, "Number of output channels. Defaults to input.");
+DEFINE_int32(out_sample_rate, 0,
+             "Output sample rate in Hz. Defaults to input.");
 DEFINE_string(mic_positions, "",
     "Space delimited cartesian coordinates of microphones in meters. "
     "The coordinates of each point are contiguous. "
     "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
-DEFINE_double(
-    target_angle_degrees,
-    90,
-    "The azimuth of the target in degrees. Only applies to beamforming.");
+DEFINE_double(target_angle_degrees, 90, "The azimuth of the target in radians");
 
 DEFINE_bool(aec, false, "Enable echo cancellation.");
 DEFINE_bool(agc, false, "Enable automatic gain control.");
 DEFINE_bool(hpf, false, "Enable high-pass filtering.");
 DEFINE_bool(ns, false, "Enable noise suppression.");
 DEFINE_bool(ts, false, "Enable transient suppression.");
 DEFINE_bool(bf, false, "Enable beamforming.");
 DEFINE_bool(ie, false, "Enable intelligibility enhancer.");
 DEFINE_bool(all, false, "Enable all components.");
 
 DEFINE_int32(ns_level, -1, "Noise suppression level [0 - 3].");
 
 DEFINE_bool(perf, false, "Enable performance tests.");
 
 namespace webrtc {
 namespace {
 
 const int kChunksPerSecond = 100;
 const char kUsage[] =
     "Command-line tool to run audio processing on WAV files. Accepts either\n"
     "an input capture WAV file or protobuf debug dump and writes to an output\n"
     "WAV file.\n"
     "\n"
     "All components are disabled by default. If any bi-directional components\n"
     "are enabled, only debug dump files are permitted.";
 
+// Returns a StreamConfig corresponding to wav_file if it's non-nullptr.
+// Otherwise returns a default initialized StreamConfig.
+StreamConfig MakeStreamConfig(const WavFile* wav_file) {
+  if (wav_file) {
+    return {wav_file->sample_rate(), wav_file->num_channels()};
+  }
+  return {};
+}
+
 }  // namespace
 
 int main(int argc, char* argv[]) {
   google::SetUsageMessage(kUsage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   if (!((FLAGS_i.empty()) ^ (FLAGS_dump.empty()))) {
     fprintf(stderr,
             "An input file must be specified with either -i or -dump.\n");
     return 1;
   }
-  if (FLAGS_dump.empty() && (FLAGS_aec || FLAGS_ie)) {
-    fprintf(stderr, "-aec and -ie require a -dump file.\n");
-    return 1;
-  }
-  if (FLAGS_ie) {
-    fprintf(stderr,
-            "FIXME(ajm): The intelligibility enhancer output is not dumped.\n");
+  if (!FLAGS_dump.empty()) {
+    fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
     return 1;
   }
 
   test::TraceToStderr trace_to_stderr(true);
+  WavReader in_file(FLAGS_i);
+  // If the output format is uninitialized, use the input format.
+  const int out_channels =
+      FLAGS_out_channels ? FLAGS_out_channels : in_file.num_channels();
+  const int out_sample_rate =
+      FLAGS_out_sample_rate ? FLAGS_out_sample_rate : in_file.sample_rate();
+  WavWriter out_file(FLAGS_o, out_sample_rate, out_channels);
+
   Config config;
+  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
+  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
+
   if (FLAGS_bf || FLAGS_all) {
-    if (FLAGS_mic_positions.empty()) {
-      fprintf(stderr, "-mic_positions must be specified when -bf is used.\n");
-      return 1;
-    }
+    const size_t num_mics = in_file.num_channels();
+    const std::vector<Point> array_geometry =
+        ParseArrayGeometry(FLAGS_mic_positions, num_mics);
+    RTC_CHECK_EQ(array_geometry.size(), num_mics);
+
     config.Set<Beamforming>(new Beamforming(
-        true, ParseArrayGeometry(FLAGS_mic_positions),
+        true, array_geometry,
         SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f,
                         1.f)));
   }
-  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
-  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
 
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
-  RTC_CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
+  if (!FLAGS_dump.empty()) {
+    RTC_CHECK_EQ(kNoErr,
+                 ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
+  } else if (FLAGS_aec) {
+    fprintf(stderr, "-aec requires a -dump file.\n");
+    return -1;
+  }
+  bool process_reverse = !FLAGS_i_rev.empty();
   RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
+  RTC_CHECK_EQ(kNoErr,
+               ap->gain_control()->set_mode(GainControl::kFixedDigital));
   RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
   RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
-  if (FLAGS_ns_level != -1) {
+  if (FLAGS_ns_level != -1)
     RTC_CHECK_EQ(kNoErr,
                  ap->noise_suppression()->set_level(
                      static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
   }
   ap->set_stream_key_pressed(FLAGS_ts);
 
-  rtc::scoped_ptr<AudioFileProcessor> processor;
-  auto out_file = rtc_make_scoped_ptr(
-      new WavWriter(FLAGS_o, FLAGS_out_sample_rate, FLAGS_out_channels));
-  std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl;
-  if (FLAGS_dump.empty()) {
-    auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
-    std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl;
-    processor.reset(
-        new WavFileProcessor(ap.Pass(), in_file.Pass(), out_file.Pass()));
+  printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+         FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
+  printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+         FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
 
-  } else {
-    processor.reset(new AecDumpFileProcessor(
-        ap.Pass(), fopen(FLAGS_dump.c_str(), "rb"), out_file.Pass()));
+  ChannelBuffer<float> in_buf(
+      rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
+      in_file.num_channels());
+  ChannelBuffer<float> out_buf(
+      rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
+      out_file.num_channels());
+
+  std::vector<float> in_interleaved(in_buf.size());
+  std::vector<float> out_interleaved(out_buf.size());
+
+  rtc::scoped_ptr<WavReader> in_rev_file;
+  rtc::scoped_ptr<WavWriter> out_rev_file;
+  rtc::scoped_ptr<ChannelBuffer<float>> in_rev_buf;
+  rtc::scoped_ptr<ChannelBuffer<float>> out_rev_buf;
+  std::vector<float> in_rev_interleaved;
+  std::vector<float> out_rev_interleaved;
+  if (process_reverse) {
+    in_rev_file.reset(new WavReader(FLAGS_i_rev));
+    out_rev_file.reset(new WavWriter(FLAGS_o_rev, in_rev_file->sample_rate(),
+                                     in_rev_file->num_channels()));
+    printf("In rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+           FLAGS_i_rev.c_str(), in_rev_file->num_channels(),
+           in_rev_file->sample_rate());
+    printf("Out rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+           FLAGS_o_rev.c_str(), out_rev_file->num_channels(),
+           out_rev_file->sample_rate());
+    in_rev_buf.reset(new ChannelBuffer<float>(
+        rtc::CheckedDivExact(in_rev_file->sample_rate(), kChunksPerSecond),
+        in_rev_file->num_channels()));
+    in_rev_interleaved.resize(in_rev_buf->size());
+    out_rev_buf.reset(new ChannelBuffer<float>(
+        rtc::CheckedDivExact(out_rev_file->sample_rate(), kChunksPerSecond),
+        out_rev_file->num_channels()));
+    out_rev_interleaved.resize(out_rev_buf->size());
   }
 
+  TickTime processing_start_time;
+  TickInterval accumulated_time;
   int num_chunks = 0;
-  while (processor->ProcessChunk()) {
+
+  const auto input_config = MakeStreamConfig(&in_file);
+  const auto output_config = MakeStreamConfig(&out_file);
+  const auto reverse_input_config = MakeStreamConfig(in_rev_file.get());
+  const auto reverse_output_config = MakeStreamConfig(out_rev_file.get());
+
+  while (in_file.ReadSamples(in_interleaved.size(),
+                             &in_interleaved[0]) == in_interleaved.size()) {
+    // Have logs display the file time rather than wallclock time.
     trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond);
-    ++num_chunks;
+    FloatS16ToFloat(&in_interleaved[0], in_interleaved.size(),
+                    &in_interleaved[0]);
+    Deinterleave(&in_interleaved[0], in_buf.num_frames(),
+                 in_buf.num_channels(), in_buf.channels());
+    if (process_reverse) {
+      in_rev_file->ReadSamples(in_rev_interleaved.size(),
+                               in_rev_interleaved.data());
+      FloatS16ToFloat(in_rev_interleaved.data(), in_rev_interleaved.size(),
+                      in_rev_interleaved.data());
+      Deinterleave(in_rev_interleaved.data(), in_rev_buf->num_frames(),
+                   in_rev_buf->num_channels(), in_rev_buf->channels());
+    }
+
+    if (FLAGS_perf) {
+      processing_start_time = TickTime::Now();
+    }
+    RTC_CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config,
+                                           output_config, out_buf.channels()));
+    if (process_reverse) {
+      RTC_CHECK_EQ(kNoErr, ap->ProcessReverseStream(
+                               in_rev_buf->channels(), reverse_input_config,
+                               reverse_output_config, out_rev_buf->channels()));
+    }
+    if (FLAGS_perf) {
+      accumulated_time += TickTime::Now() - processing_start_time;
+    }
+
+    Interleave(out_buf.channels(), out_buf.num_frames(),
+               out_buf.num_channels(), &out_interleaved[0]);
+    FloatToFloatS16(&out_interleaved[0], out_interleaved.size(),
+                    &out_interleaved[0]);
+    out_file.WriteSamples(&out_interleaved[0], out_interleaved.size());
+    if (process_reverse) {
+      Interleave(out_rev_buf->channels(), out_rev_buf->num_frames(),
+                 out_rev_buf->num_channels(), out_rev_interleaved.data());
+      FloatToFloatS16(out_rev_interleaved.data(), out_rev_interleaved.size(),
+                      out_rev_interleaved.data());
+      out_rev_file->WriteSamples(out_rev_interleaved.data(),
+                                 out_rev_interleaved.size());
+    }
+    num_chunks++;
   }
-
   if (FLAGS_perf) {
-    const auto& proc_time = processor->proc_time();
-    int64_t exec_time_us = proc_time.sum.Microseconds();
-    printf(
-        "\nExecution time: %.3f s, File time: %.2f s\n"
-        "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n",
-        exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond,
-        exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(),
-        1.f * proc_time.min.Microseconds());
+    int64_t execution_time_ms = accumulated_time.Milliseconds();
+    printf("\nExecution time: %.3f s\nFile time: %.2f s\n"
+           "Time per chunk: %.3f ms\n",
+           execution_time_ms * 0.001f, num_chunks * 1.f / kChunksPerSecond,
+           execution_time_ms * 1.f / num_chunks);
   }
-
   return 0;
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::main(argc, argv);
 }