diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
index b14b620..8dae7d6 100644
--- a/webrtc/common_audio/wav_file.cc
+++ b/webrtc/common_audio/wav_file.cc
@@ -37,17 +37,9 @@
   FILE* file_;
 };
 
-std::string WavFile::FormatAsString() const {
-  std::ostringstream s;
-  s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
-    << ", Duration: "
-    << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
-  return s.str();
-}
-
 WavReader::WavReader(const std::string& filename)
     : file_handle_(fopen(filename.c_str(), "rb")) {
-  RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
+  RTC_CHECK(file_handle_ && "Could not open wav file for reading.");
 
   ReadableWavFile readable(file_handle_);
   WavFormat format;
@@ -104,7 +96,7 @@
       num_channels_(num_channels),
       num_samples_(0),
       file_handle_(fopen(filename.c_str(), "wb")) {
-  RTC_CHECK(file_handle_) << "Could not open wav file for writing.";
+  RTC_CHECK(file_handle_ && "Could not open wav file for writing.");
   RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
                                kBytesPerSample, num_samples_));
 
diff --git a/webrtc/common_audio/wav_file.h b/webrtc/common_audio/wav_file.h
index 42b0618..2eadd3f 100644
--- a/webrtc/common_audio/wav_file.h
+++ b/webrtc/common_audio/wav_file.h
@@ -29,9 +29,6 @@
   virtual int sample_rate() const = 0;
   virtual int num_channels() const = 0;
   virtual uint32_t num_samples() const = 0;
-
-  // Returns a human-readable string containing the audio format.
-  std::string FormatAsString() const;
 };
 
 // Simple C++ class for writing 16-bit PCM WAV files. All error handling is
diff --git a/webrtc/modules/audio_processing/audio_processing_tests.gypi b/webrtc/modules/audio_processing/audio_processing_tests.gypi
index b301b00..0314c69 100644
--- a/webrtc/modules/audio_processing/audio_processing_tests.gypi
+++ b/webrtc/modules/audio_processing/audio_processing_tests.gypi
@@ -12,13 +12,10 @@
       'target_name': 'audioproc_test_utils',
       'type': 'static_library',
       'dependencies': [
-        'audioproc_debug_proto',
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       ],
       'sources': [
-        'test/audio_file_processor.cc',
-        'test/audio_file_processor.h',
         'test/test_utils.cc',
         'test/test_utils.h',
       ],
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
deleted file mode 100644
index ca244d5..0000000
--- a/webrtc/modules/audio_processing/test/audio_file_processor.cc
+++ /dev/null
@@ -1,177 +0,0 @@
-/*
- *  Copyright (c) 2015 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 "webrtc/modules/audio_processing/test/audio_file_processor.h"
-
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
-
-using rtc::scoped_ptr;
-using rtc::CheckedDivExact;
-using std::vector;
-using webrtc::audioproc::Event;
-using webrtc::audioproc::Init;
-using webrtc::audioproc::ReverseStream;
-using webrtc::audioproc::Stream;
-
-namespace webrtc {
-namespace {
-
-// Returns a StreamConfig corresponding to file.
-StreamConfig GetStreamConfig(const WavFile& file) {
-  return StreamConfig(file.sample_rate(), file.num_channels());
-}
-
-// Returns a ChannelBuffer corresponding to file.
-ChannelBuffer<float> GetChannelBuffer(const WavFile& file) {
-  return ChannelBuffer<float>(
-      CheckedDivExact(file.sample_rate(), AudioFileProcessor::kChunksPerSecond),
-      file.num_channels());
-}
-
-}  // namespace
-
-WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
-                                   scoped_ptr<WavReader> in_file,
-                                   scoped_ptr<WavWriter> out_file)
-    : ap_(ap.Pass()),
-      in_buf_(GetChannelBuffer(*in_file)),
-      out_buf_(GetChannelBuffer(*out_file)),
-      input_config_(GetStreamConfig(*in_file)),
-      output_config_(GetStreamConfig(*out_file)),
-      buffer_reader_(in_file.Pass()),
-      buffer_writer_(out_file.Pass()) {}
-
-bool WavFileProcessor::ProcessChunk() {
-  if (!buffer_reader_.Read(&in_buf_)) {
-    return false;
-  }
-  {
-    const auto st = ScopedTimer(mutable_proc_time());
-    RTC_CHECK_EQ(kNoErr,
-                 ap_->ProcessStream(in_buf_.channels(), input_config_,
-                                    output_config_, out_buf_.channels()));
-  }
-  buffer_writer_.Write(out_buf_);
-  return true;
-}
-
-AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,
-                                           FILE* dump_file,
-                                           scoped_ptr<WavWriter> out_file)
-    : ap_(ap.Pass()),
-      dump_file_(dump_file),
-      out_buf_(GetChannelBuffer(*out_file)),
-      output_config_(GetStreamConfig(*out_file)),
-      buffer_writer_(out_file.Pass()) {
-  RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
-}
-
-AecDumpFileProcessor::~AecDumpFileProcessor() {
-  fclose(dump_file_);
-}
-
-bool AecDumpFileProcessor::ProcessChunk() {
-  Event event_msg;
-
-  // Continue until we process our first Stream message.
-  do {
-    if (!ReadMessageFromFile(dump_file_, &event_msg)) {
-      return false;
-    }
-
-    if (event_msg.type() == Event::INIT) {
-      RTC_CHECK(event_msg.has_init());
-      HandleMessage(event_msg.init());
-
-    } else if (event_msg.type() == Event::STREAM) {
-      RTC_CHECK(event_msg.has_stream());
-      HandleMessage(event_msg.stream());
-
-    } else if (event_msg.type() == Event::REVERSE_STREAM) {
-      RTC_CHECK(event_msg.has_reverse_stream());
-      HandleMessage(event_msg.reverse_stream());
-    }
-  } while (event_msg.type() != Event::STREAM);
-
-  return true;
-}
-
-void AecDumpFileProcessor::HandleMessage(const Init& msg) {
-  RTC_CHECK(msg.has_sample_rate());
-  RTC_CHECK(msg.has_num_input_channels());
-  RTC_CHECK(msg.has_num_reverse_channels());
-
-  in_buf_.reset(new ChannelBuffer<float>(
-      CheckedDivExact(msg.sample_rate(), kChunksPerSecond),
-      msg.num_input_channels()));
-  const int reverse_sample_rate = msg.has_reverse_sample_rate()
-                                      ? msg.reverse_sample_rate()
-                                      : msg.sample_rate();
-  reverse_buf_.reset(new ChannelBuffer<float>(
-      CheckedDivExact(reverse_sample_rate, kChunksPerSecond),
-      msg.num_reverse_channels()));
-  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
-  reverse_config_ =
-      StreamConfig(reverse_sample_rate, msg.num_reverse_channels());
-
-  const ProcessingConfig config = {
-      {input_config_, output_config_, reverse_config_, reverse_config_}};
-  RTC_CHECK_EQ(kNoErr, ap_->Initialize(config));
-}
-
-void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
-  RTC_CHECK(!msg.has_input_data());
-  RTC_CHECK_EQ(in_buf_->num_channels(), msg.input_channel_size());
-
-  for (int i = 0; i < msg.input_channel_size(); ++i) {
-    RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
-                 msg.input_channel(i).size());
-    std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
-                msg.input_channel(i).size());
-  }
-  {
-    const auto st = ScopedTimer(mutable_proc_time());
-    RTC_CHECK_EQ(kNoErr, ap_->set_stream_delay_ms(msg.delay()));
-    ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
-    if (msg.has_keypress()) {
-      ap_->set_stream_key_pressed(msg.keypress());
-    }
-    RTC_CHECK_EQ(kNoErr,
-                 ap_->ProcessStream(in_buf_->channels(), input_config_,
-                                    output_config_, out_buf_.channels()));
-  }
-
-  buffer_writer_.Write(out_buf_);
-}
-
-void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
-  RTC_CHECK(!msg.has_data());
-  RTC_CHECK_EQ(reverse_buf_->num_channels(), msg.channel_size());
-
-  for (int i = 0; i < msg.channel_size(); ++i) {
-    RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
-                 msg.channel(i).size());
-    std::memcpy(reverse_buf_->channels()[i], msg.channel(i).data(),
-                msg.channel(i).size());
-  }
-  {
-    const auto st = ScopedTimer(mutable_proc_time());
-    // TODO(ajm): This currently discards the processed output, which is needed
-    // for e.g. intelligibility enhancement.
-    RTC_CHECK_EQ(kNoErr, ap_->ProcessReverseStream(
-                             reverse_buf_->channels(), reverse_config_,
-                             reverse_config_, reverse_buf_->channels()));
-  }
-}
-
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.h b/webrtc/modules/audio_processing/test/audio_file_processor.h
deleted file mode 100644
index a3153b2..0000000
--- a/webrtc/modules/audio_processing/test/audio_file_processor.h
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- *  Copyright (c) 2015 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
-
-#include <algorithm>
-#include <limits>
-#include <vector>
-
-#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/test_utils.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
-#else
-#include "webrtc/audio_processing/debug.pb.h"
-#endif
-
-namespace webrtc {
-
-// Holds a few statistics about a series of TickIntervals.
-struct TickIntervalStats {
-  TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
-  TickInterval sum;
-  TickInterval max;
-  TickInterval min;
-};
-
-// Interface for processing an input file with an AudioProcessing instance and
-// dumping the results to an output file.
-class AudioFileProcessor {
- public:
-  static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
-
-  virtual ~AudioFileProcessor() {}
-
-  // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
-  // writes to the output file.
-  virtual bool ProcessChunk() = 0;
-
-  // Returns the execution time of all AudioProcessing calls.
-  const TickIntervalStats& proc_time() const { return proc_time_; }
-
- protected:
-  // RAII class for execution time measurement. Updates the provided
-  // TickIntervalStats based on the time between ScopedTimer creation and
-  // leaving the enclosing scope.
-  class ScopedTimer {
-   public:
-    explicit ScopedTimer(TickIntervalStats* proc_time)
-        : proc_time_(proc_time), start_time_(TickTime::Now()) {}
-
-    ~ScopedTimer() {
-      TickInterval interval = TickTime::Now() - start_time_;
-      proc_time_->sum += interval;
-      proc_time_->max = std::max(proc_time_->max, interval);
-      proc_time_->min = std::min(proc_time_->min, interval);
-    }
-
-   private:
-    TickIntervalStats* const proc_time_;
-    TickTime start_time_;
-  };
-
-  TickIntervalStats* mutable_proc_time() { return &proc_time_; }
-
- private:
-  TickIntervalStats proc_time_;
-};
-
-// Used to read from and write to WavFile objects.
-class WavFileProcessor final : public AudioFileProcessor {
- public:
-  // Takes ownership of all parameters.
-  WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
-                   rtc::scoped_ptr<WavReader> in_file,
-                   rtc::scoped_ptr<WavWriter> out_file);
-  virtual ~WavFileProcessor() {}
-
-  // Processes one chunk from the WAV input and writes to the WAV output.
-  bool ProcessChunk() override;
-
- private:
-  rtc::scoped_ptr<AudioProcessing> ap_;
-
-  ChannelBuffer<float> in_buf_;
-  ChannelBuffer<float> out_buf_;
-  const StreamConfig input_config_;
-  const StreamConfig output_config_;
-  ChannelBufferWavReader buffer_reader_;
-  ChannelBufferWavWriter buffer_writer_;
-};
-
-// Used to read from an aecdump file and write to a WavWriter.
-class AecDumpFileProcessor final : public AudioFileProcessor {
- public:
-  // Takes ownership of all parameters.
-  AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
-                       FILE* dump_file,
-                       rtc::scoped_ptr<WavWriter> out_file);
-
-  virtual ~AecDumpFileProcessor();
-
-  // Processes messages from the aecdump file until the first Stream message is
-  // completed. Passes other data from the aecdump messages as appropriate.
-  bool ProcessChunk() override;
-
- private:
-  void HandleMessage(const webrtc::audioproc::Init& msg);
-  void HandleMessage(const webrtc::audioproc::Stream& msg);
-  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
-
-  rtc::scoped_ptr<AudioProcessing> ap_;
-  FILE* dump_file_;
-
-  rtc::scoped_ptr<ChannelBuffer<float>> in_buf_;
-  rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_;
-  ChannelBuffer<float> out_buf_;
-  StreamConfig input_config_;
-  StreamConfig reverse_config_;
-  const StreamConfig output_config_;
-  ChannelBufferWavWriter buffer_writer_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 3f1dc37..27f69b3 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -9,7 +9,6 @@
  */
 
 #include <stdio.h>
-#include <iostream>
 #include <sstream>
 #include <string>
 
@@ -19,28 +18,26 @@
 #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.");
@@ -67,6 +64,15 @@
     "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[]) {
@@ -78,75 +84,160 @@
             "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;
-  if (FLAGS_bf || FLAGS_all) {
-    if (FLAGS_mic_positions.empty()) {
-      fprintf(stderr, "-mic_positions must be specified when -bf is used.\n");
-      return 1;
-    }
-    config.Set<Beamforming>(new Beamforming(
-        true, ParseArrayGeometry(FLAGS_mic_positions),
-        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));
 
+  if (FLAGS_bf || FLAGS_all) {
+    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, array_geometry,
+        SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f,
+                        1.f)));
+  }
+
   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;
 }
 
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 1383bbe..fdfaab0 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -636,8 +636,8 @@
         }
 
         if (!raw_output) {
-          // The WAV file needs to be reset every time, because it can't change
-          // its sample rate or number of channels.
+          // The WAV file needs to be reset every time, because it cant change
+          // it's sample rate or number of channels.
           output_wav_file.reset(new WavWriter(out_filename + ".wav",
                                               output_sample_rate,
                                               msg.num_output_channels()));
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 47bd314..1b9ac3c 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -31,35 +31,6 @@
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
-ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
-    : file_(file.Pass()) {}
-
-bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
-  RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
-  interleaved_.resize(buffer->size());
-  if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
-      interleaved_.size()) {
-    return false;
-  }
-
-  FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
-  Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
-               buffer->channels());
-  return true;
-}
-
-ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
-    : file_(file.Pass()) {}
-
-void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
-  RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
-  interleaved_.resize(buffer.size());
-  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
-             &interleaved_[0]);
-  FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
-  file_->WriteSamples(&interleaved_[0], interleaved_.size());
-}
-
 void WriteIntData(const int16_t* data,
                   size_t length,
                   WavWriter* wav_file,
@@ -121,32 +92,28 @@
     case 2:
       return AudioProcessing::kStereo;
     default:
-      RTC_CHECK(false);
+      assert(false);
       return AudioProcessing::kMono;
   }
 }
 
-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
+                                      size_t num_mics) {
   const std::vector<float> values = ParseList<float>(mic_positions);
-  const size_t num_mics =
-      rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
-  RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";
+  RTC_CHECK_EQ(values.size(), 3 * num_mics)
+      << "Could not parse mic_positions or incorrect number of points.";
 
   std::vector<Point> result;
   result.reserve(num_mics);
   for (size_t i = 0; i < values.size(); i += 3) {
-    result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
+    double x = values[i + 0];
+    double y = values[i + 1];
+    double z = values[i + 2];
+    result.push_back(Point(x, y, z));
   }
 
   return result;
 }
 
-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
-                                      size_t num_mics) {
-  std::vector<Point> result = ParseArrayGeometry(mic_positions);
-  RTC_CHECK_EQ(result.size(), num_mics)
-      << "Could not parse mic_positions or incorrect number of points.";
-  return result;
-}
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index 93a0138..75e4239 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -43,35 +43,6 @@
   RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
 };
 
-// Reads ChannelBuffers from a provided WavReader.
-class ChannelBufferWavReader final {
- public:
-  explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file);
-
-  // Reads data from the file according to the |buffer| format. Returns false if
-  // a full buffer can't be read from the file.
-  bool Read(ChannelBuffer<float>* buffer);
-
- private:
-  rtc::scoped_ptr<WavReader> file_;
-  std::vector<float> interleaved_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader);
-};
-
-// Writes ChannelBuffers to a provided WavWriter.
-class ChannelBufferWavWriter final {
- public:
-  explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file);
-  void Write(const ChannelBuffer<float>& buffer);
-
- private:
-  rtc::scoped_ptr<WavWriter> file_;
-  std::vector<float> interleaved_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter);
-};
-
 void WriteIntData(const int16_t* data,
                   size_t length,
                   WavWriter* wav_file,
@@ -147,9 +118,6 @@
 std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
                                       size_t num_mics);
 
-// Same as above, but without the num_mics check for when it isn't available.
-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
diff --git a/webrtc/system_wrappers/include/tick_util.h b/webrtc/system_wrappers/include/tick_util.h
index 46d62cc..f839ff6 100644
--- a/webrtc/system_wrappers/include/tick_util.h
+++ b/webrtc/system_wrappers/include/tick_util.h
@@ -83,7 +83,6 @@
 class TickInterval {
  public:
   TickInterval();
-  explicit TickInterval(int64_t interval);
 
   int64_t Milliseconds() const;
   int64_t Microseconds() const;
@@ -104,6 +103,8 @@
   friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
 
  private:
+  explicit TickInterval(int64_t interval);
+
   friend class TickTime;
   friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);