Changed ramping functionality of the AudioMixer.

webrtc/modules/audio_mixer/audio_frame_manipulator.cc

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
index b8274d3f62d0443165492a5bd1afd7b0fda2b142..6a1e2b130602daed4d447372138d3978df15449b 100644
--- a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
@@ -15,22 +15,6 @@
 #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;
@@ -42,24 +26,21 @@ uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
   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(AudioFrame* audio_frame, float current, float target) {
+  RTC_DCHECK_GE(current, 0.);
+  RTC_DCHECK_GE(target, 0.);
 
-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]);
+  size_t samples =
+      audio_frame->samples_per_channel_ * audio_frame->num_channels_;
+  float fraction = 1.0 / samples;
+
+  for (size_t i = 0; i < samples; i++) {

[hlundin-webrtc, 2016/10/07 15:03:10]

I'm not an expert on performance, but I think it would be better to implement
this as a nested for loop, rather than have a division in the critical path.

In essence:

size_t samples = audio_frame->samples_per_channel_;
float increment = (target - current) / samples;
float gain = current;
for (size_t i = 0; i < samples; ++i) {
  for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
    audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain;
  }
  gain += increment;
}

WDYT?

[aleloi, 2016/10/10 10:58:17]

Sure! Your suggestion accesses the data in the same order and avoids division
inside the loop. I don't know if it's faster, but it shouldn't be slower!

+    // If the audio is interleaved of several channels, we want to
+    // apply the same gain change to the nth sample of every channel.
+    audio_frame->data_[i] *=
+        current +
+        (target - current) * (i / audio_frame->samples_per_channel_) * fraction;
   }
-  memset(&audio_frame->data_[kRampSize], 0,
-         (audio_frame->samples_per_channel_ - kRampSize) *
-             sizeof(audio_frame->data_[0]));
 }
 
 void RemixFrame(AudioFrame* frame, size_t target_number_of_channels) {
@@ -71,5 +52,4 @@ void RemixFrame(AudioFrame* frame, size_t target_number_of_channels) {
     AudioFrameOperations::StereoToMono(frame);
   }
 }
-
 }  // namespace webrtc