Index: webrtc/modules/audio_mixer/audio_frame_manipulator.cc |
diff --git a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..aafeedaeda9748100fe98b49dba72c516234ba19 |
--- /dev/null |
+++ b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc |
@@ -0,0 +1,62 @@ |
+/* |
+ * 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/modules/audio_mixer/audio_frame_manipulator.h" |
+#include "webrtc/modules/include/module_common_types.h" |
+#include "webrtc/typedefs.h" |
+ |
+namespace webrtc { |
+namespace { |
+// Linear ramping over 80 samples. |
+// TODO(hellner): ramp using fix point? |
+const float kRampArray[] = { |
+ 0.0000f, 0.0127f, 0.0253f, 0.0380f, 0.0506f, 0.0633f, 0.0759f, 0.0886f, |
+ 0.1013f, 0.1139f, 0.1266f, 0.1392f, 0.1519f, 0.1646f, 0.1772f, 0.1899f, |
+ 0.2025f, 0.2152f, 0.2278f, 0.2405f, 0.2532f, 0.2658f, 0.2785f, 0.2911f, |
+ 0.3038f, 0.3165f, 0.3291f, 0.3418f, 0.3544f, 0.3671f, 0.3797f, 0.3924f, |
+ 0.4051f, 0.4177f, 0.4304f, 0.4430f, 0.4557f, 0.4684f, 0.4810f, 0.4937f, |
+ 0.5063f, 0.5190f, 0.5316f, 0.5443f, 0.5570f, 0.5696f, 0.5823f, 0.5949f, |
+ 0.6076f, 0.6203f, 0.6329f, 0.6456f, 0.6582f, 0.6709f, 0.6835f, 0.6962f, |
+ 0.7089f, 0.7215f, 0.7342f, 0.7468f, 0.7595f, 0.7722f, 0.7848f, 0.7975f, |
+ 0.8101f, 0.8228f, 0.8354f, 0.8481f, 0.8608f, 0.8734f, 0.8861f, 0.8987f, |
+ 0.9114f, 0.9241f, 0.9367f, 0.9494f, 0.9620f, 0.9747f, 0.9873f, 1.0000f}; |
+const size_t kRampSize = sizeof(kRampArray) / sizeof(kRampArray[0]); |
+} // namespace |
+ |
+uint32_t NewMixerCalculateEnergy(const AudioFrame& audio_frame) { |
+ uint32_t energy = 0; |
+ for (size_t position = 0; position < audio_frame.samples_per_channel_; |
+ position++) { |
+ // TODO(andrew): this can easily overflow. |
+ energy += audio_frame.data_[position] * audio_frame.data_[position]; |
+ } |
+ return energy; |
+} |
+ |
+void NewMixerRampIn(AudioFrame* audio_frame) { |
+ assert(kRampSize <= audio_frame->samples_per_channel_); |
+ for (size_t i = 0; i < kRampSize; i++) { |
+ audio_frame->data_[i] = |
+ static_cast<int16_t>(kRampArray[i] * audio_frame->data_[i]); |
+ } |
+} |
+ |
+void NewMixerRampOut(AudioFrame* audio_frame) { |
+ assert(kRampSize <= audio_frame->samples_per_channel_); |
+ for (size_t i = 0; i < kRampSize; i++) { |
+ const size_t kRampPos = kRampSize - 1 - i; |
+ audio_frame->data_[i] = |
+ static_cast<int16_t>(kRampArray[kRampPos] * audio_frame->data_[i]); |
+ } |
+ memset(&audio_frame->data_[kRampSize], 0, |
+ (audio_frame->samples_per_channel_ - kRampSize) * |
+ sizeof(audio_frame->data_[0])); |
+} |
+} // namespace webrtc |