Remove <(webrtc_root) from source file entries.

This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.

Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).

I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.

BUG=4185
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37859004

Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/tools/agc/test_utils.cc

@@ -0,0 +1,63 @@
+/*
+ *  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 "webrtc/tools/agc/test_utils.h"
+
+#include <cmath>
+
+#include <algorithm>
+
+#include "webrtc/modules/interface/module_common_types.h"
+
+namespace webrtc {
+
+float MicLevel2Gain(int gain_range_db, int level) {
+  return (level - 127.0f) / 128.0f * gain_range_db / 2;
+}
+
+float Db2Linear(float db) {
+  return powf(10.0f, db / 20.0f);
+}
+
+void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
+  const int frame_length = frame->samples_per_channel_ * frame->num_channels_;
+  // Smooth the transition between gain levels across the frame.
+  float smoothed_gain = last_gain;
+  float gain_step = (gain - last_gain) / (frame_length - 1);
+  for (int i = 0; i < frame_length; ++i) {
+    smoothed_gain += gain_step;
+    float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
+    sample = std::max(std::min(32767.0f, sample), -32768.0f);
+    frame->data_[i] = static_cast<int16_t>(sample);
+  }
+}
+
+void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame) {
+  ApplyGainLinear(Db2Linear(gain_db), Db2Linear(last_gain_db), frame);
+}
+
+void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
+                 AudioFrame* frame) {
+  assert(mic_level >= 0 && mic_level <= 255);
+  assert(last_mic_level >= 0 && last_mic_level <= 255);
+  ApplyGain(MicLevel2Gain(gain_range_db, mic_level),
+            MicLevel2Gain(gain_range_db, last_mic_level),
+            frame);
+}
+
+void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
+                 AudioFrame* frame) {
+  assert(mic_level >= 0 && mic_level <= 255);
+  assert(last_mic_level >= 0 && last_mic_level <= 255);
+  ApplyGain(gain_map[mic_level], gain_map[last_mic_level], frame);
+}
+
+}  // namespace webrtc
+