Changed ramping functionality of the AudioMixer.

webrtc/modules/audio_mixer/audio_frame_manipulator.cc

 /*
  *  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/base/checks.h"
 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.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 AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
   uint32_t energy = 0;
   for (size_t position = 0; position < audio_frame.samples_per_channel_;
        position++) {
     // TODO(aleloi): This can overflow. Convert to floats.
     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 Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
+  RTC_DCHECK(audio_frame);
+  RTC_DCHECK_GE(start_gain, 0.0f);
+  RTC_DCHECK_GE(target_gain, 0.0f);
+
+  size_t samples = audio_frame->samples_per_channel_;
+  RTC_DCHECK_LT(0u, samples);
+  float increment = (target_gain - start_gain) / samples;
+  float gain = start_gain;
+  for (size_t i = 0; i < samples; ++i) {
+    // If the audio is interleaved of several channels, we want to
+    // apply the same gain change to the ith sample of every channel.
+    for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
+      audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain;
+    }
+    gain += increment;
   }
 }
 
-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]));
-}
-
 void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
-  RTC_DCHECK_GE(target_number_of_channels, static_cast<size_t>(1));
-  RTC_DCHECK_LE(target_number_of_channels, static_cast<size_t>(2));
+  RTC_DCHECK_GE(target_number_of_channels, 1u);
+  RTC_DCHECK_LE(target_number_of_channels, 2u);
   if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
     AudioFrameOperations::MonoToStereo(frame);
   } else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
     AudioFrameOperations::StereoToMono(frame);
   }
 }
-
 }  // namespace webrtc