Remove AgcManager.

webrtc/tools/agc/agc_test.cc

Index: webrtc/tools/agc/agc_test.cc
diff --git a/webrtc/tools/agc/agc_test.cc b/webrtc/tools/agc/agc_test.cc
deleted file mode 100644
index e8c4eb8884a09cd1cf1d779fb6b9f7c23b413cbb..0000000000000000000000000000000000000000
--- a/webrtc/tools/agc/agc_test.cc
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- *  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.
- */
-
-#include <cmath>
-#include <cstdio>
-
-#include <algorithm>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-#include "webrtc/tools/agc/agc_manager.h"
-#include "webrtc/tools/agc/test_utils.h"
-#include "webrtc/voice_engine/mock/fake_voe_external_media.h"
-#include "webrtc/voice_engine/mock/mock_voe_volume_control.h"
-
-DEFINE_string(in, "in.pcm", "input filename");
-DEFINE_string(out, "out.pcm", "output filename");
-DEFINE_int32(rate, 16000, "sample rate in Hz");
-DEFINE_int32(channels, 1, "number of channels");
-DEFINE_int32(level, -18, "target level in RMS dBFs [-100, 0]");
-DEFINE_bool(limiter, true, "enable a limiter for the compression stage");
-DEFINE_int32(cmp_level, 2, "target level in dBFs for the compression stage");
-DEFINE_int32(mic_gain, 80, "range of gain provided by the virtual mic in dB");
-DEFINE_int32(gain_offset, 0,
-             "an amount (in dB) to add to every entry in the gain map");
-DEFINE_string(gain_file, "",
-    "filename providing a mic gain mapping. The file should be text containing "
-    "a (floating-point) gain entry in dBFs per line corresponding to levels "
-    "from 0 to 255.");
-
-using ::testing::_;
-using ::testing::ByRef;
-using ::testing::DoAll;
-using ::testing::Mock;
-using ::testing::Return;
-using ::testing::SaveArg;
-using ::testing::SetArgReferee;
-
-namespace webrtc {
-namespace {
-
-const char kUsage[] = "\nProcess an audio file to simulate an analog agc.";
-
-void ReadGainMapFromFile(FILE* file, int offset, int gain_map[256]) {
-  for (int i = 0; i < 256; ++i) {
-    float gain = 0;
-    ASSERT_EQ(1, fscanf(file, "%f", &gain));
-    gain_map[i] = std::floor(gain + 0.5);
-  }
-
-  // Adjust from dBFs to gain in dB. We assume that level 127 provides 0 dB
-  // gain. This corresponds to the interpretation in MicLevel2Gain().
-  const int midpoint = gain_map[127];
-  printf("Gain map\n");
-  for (int i = 0; i < 256; ++i) {
-    gain_map[i] += offset - midpoint;
-    if (i % 5 == 0) {
-      printf("%d: %d dB\n", i, gain_map[i]);
-    }
-  }
-}
-
-void CalculateGainMap(int gain_range_db, int offset, int gain_map[256]) {
-  printf("Gain map\n");
-  for (int i = 0; i < 256; ++i) {
-    gain_map[i] = std::floor(MicLevel2Gain(gain_range_db, i) + 0.5) + offset;
-    if (i % 5 == 0) {
-      printf("%d: %d dB\n", i, gain_map[i]);
-    }
-  }
-}
-
-void RunAgc() {
-  test::TraceToStderr trace_to_stderr(true);
-  FILE* in_file = fopen(FLAGS_in.c_str(), "rb");
-  ASSERT_TRUE(in_file != NULL);
-  FILE* out_file = fopen(FLAGS_out.c_str(), "wb");
-  ASSERT_TRUE(out_file != NULL);
-
-  int gain_map[256];
-  if (!FLAGS_gain_file.empty()) {
-    FILE* gain_file = fopen(FLAGS_gain_file.c_str(), "rt");
-    ASSERT_TRUE(gain_file != NULL);
-    ReadGainMapFromFile(gain_file, FLAGS_gain_offset, gain_map);
-    fclose(gain_file);
-  } else {
-    CalculateGainMap(FLAGS_mic_gain, FLAGS_gain_offset, gain_map);
-  }
-
-  FakeVoEExternalMedia media;
-  MockVoEVolumeControl volume;
-  Agc* agc = new Agc;
-  AudioProcessing* audioproc = AudioProcessing::Create();
-  ASSERT_TRUE(audioproc != NULL);
-  AgcManager manager(&media, &volume, agc, audioproc);
-
-  int mic_level = 128;
-  int last_mic_level = mic_level;
-  EXPECT_CALL(volume, GetMicVolume(_))
-      .WillRepeatedly(DoAll(SetArgReferee<0>(ByRef(mic_level)), Return(0)));
-  EXPECT_CALL(volume, SetMicVolume(_))
-      .WillRepeatedly(DoAll(SaveArg<0>(&mic_level), Return(0)));
-
-  manager.Enable(true);
-  ASSERT_EQ(0, agc->set_target_level_dbfs(FLAGS_level));
-  const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
-  GainControl* gctrl = audioproc->gain_control();
-  ASSERT_EQ(kNoErr, gctrl->set_target_level_dbfs(FLAGS_cmp_level));
-  ASSERT_EQ(kNoErr, gctrl->enable_limiter(FLAGS_limiter));
-
-  AudioFrame frame;
-  frame.num_channels_ = FLAGS_channels;
-  frame.sample_rate_hz_ = FLAGS_rate;
-  frame.samples_per_channel_ = FLAGS_rate / 100;
-  const size_t frame_length = frame.samples_per_channel_ * FLAGS_channels;
-  size_t sample_count = 0;
-  while (fread(frame.data_, sizeof(int16_t), frame_length, in_file) ==
-      frame_length) {
-    SimulateMic(gain_map, mic_level, last_mic_level, &frame);
-    last_mic_level = mic_level;
-    media.CallProcess(kRecordingAllChannelsMixed, frame.data_,
-                      frame.samples_per_channel_, FLAGS_rate, FLAGS_channels);
-    ASSERT_EQ(frame_length,
-              fwrite(frame.data_, sizeof(int16_t), frame_length, out_file));
-    sample_count += frame_length;
-    trace_to_stderr.SetTimeSeconds(static_cast<float>(sample_count) /
-                                   FLAGS_channels / FLAGS_rate);
-  }
-  fclose(in_file);
-  fclose(out_file);
-  EXPECT_CALL(volume, Release());
-}
-
-}  // namespace
-}  // namespace webrtc
-
-int main(int argc, char* argv[]) {
-  google::SetUsageMessage(webrtc::kUsage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::RunAgc();
-  return 0;
-}