Moved MixerAudioSource and removed audio_mixer_defines.h.

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 7ced9e29fa9e1c522c0c2de0cb3a1a9e990ec386..4cf9741f3b40625468103ff7df750eff489c0c0f 100644
--- a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
@@ -8,27 +8,13 @@
  *  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;
@@ -40,23 +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) {
+  int samples = static_cast<int>(audio_frame->samples_per_channel_ *

[hlundin-webrtc, 2016/10/07 13:18:36]

Sanity check the current and target gains?

[hlundin-webrtc, 2016/10/07 13:18:36]

int -> size_t

[aleloi, 2016/10/07 14:23:23]

Done in next CL https://codereview.webrtc.org/2398083005

+                                 audio_frame->num_channels_);
+  float fraction = 1.0 / samples;
+  for (int i = 0; i < samples; i++) {
+    audio_frame->data_[i] *= current + (target - current) * i * fraction;

[hlundin-webrtc, 2016/10/07 13:18:36]

Wait a minute. This looks like you are applying different gains for the same
sample but in different channels. For instance, in the case of ramping down from
1, in stereo, I would expect the gains to be like this:
Left:  [1.0 0.9 0.8 ... ]
Right: [1.0 0.9 0.8 ... ]

I used a .1 decrement here for simplicity, the point is that the all values for
the same time instant should get the same gain across all channels.

I realize that you are keeping the behavior of the old code, but I think that is
wrong.

[aleloi, 2016/10/07 14:23:23]

I changed the loop to the following in the next CL:

audio_frame->data_[i] *=
 current +
 (target - current) * (i / audio_frame->samples_per_channel_) * fraction;

   }
 }
 
-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]);
+void RemixFrame(AudioFrame* frame, size_t number_of_channels) {
+  RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2);

[hlundin-webrtc, 2016/10/07 13:18:36]

You get better DCHECK output if you do this in two steps:
RTC_DCHECK_GE(number_of_channels, 1);
RTC_DCHECK_LE(number_of_channels, 2);

[aleloi, 2016/10/07 14:23:23]

Done.

+  if (frame->num_channels_ == 1 && number_of_channels == 2) {
+    AudioFrameOperations::MonoToStereo(frame);
+  } else if (frame->num_channels_ == 2 && number_of_channels == 1) {
+    AudioFrameOperations::StereoToMono(frame);
   }
-  memset(&audio_frame->data_[kRampSize], 0,
-         (audio_frame->samples_per_channel_ - kRampSize) *
-             sizeof(audio_frame->data_[0]));
 }
 }  // namespace webrtc