Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.

We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.

Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc

Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.

BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/audio_buffer.cc

@@ -99,7 +99,6 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
     num_mixed_low_pass_channels_(0),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
-    is_muted_(false),
     data_(NULL),
     keyboard_data_(NULL),
     channels_(new ChannelBuffer<int16_t>(proc_samples_per_channel_,
@@ -223,7 +222,6 @@ void AudioBuffer::InitForNewData() {
   num_mixed_low_pass_channels_ = 0;
   reference_copied_ = false;
   activity_ = AudioFrame::kVadUnknown;
-  is_muted_ = false;
 }
 
 const int16_t* AudioBuffer::data(int channel) const {
@@ -307,10 +305,6 @@ AudioFrame::VADActivity AudioBuffer::activity() const {
   return activity_;
 }
 
-bool AudioBuffer::is_muted() const {
-  return is_muted_;
-}
-
 int AudioBuffer::num_channels() const {
   return num_proc_channels_;
 }
@@ -336,9 +330,6 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
   assert(frame->samples_per_channel_ ==  proc_samples_per_channel_);
   InitForNewData();
   activity_ = frame->vad_activity_;
-  if (frame->energy_ == 0) {
-    is_muted_ = true;
-  }
 
   if (num_proc_channels_ == 1) {
     // We can get away with a pointer assignment in this case.

webrtc/modules/audio_processing/audio_buffer.h

@@ -71,8 +71,6 @@ class AudioBuffer {
   void set_activity(AudioFrame::VADActivity activity);
   AudioFrame::VADActivity activity() const;
 
-  bool is_muted() const;
-
   // Use for int16 interleaved data.
   void DeinterleaveFrom(AudioFrame* audioFrame);
   void InterleaveTo(AudioFrame* audioFrame) const;
@@ -106,7 +104,6 @@ class AudioBuffer {
   int num_mixed_low_pass_channels_;
   bool reference_copied_;
   AudioFrame::VADActivity activity_;
-  bool is_muted_;
 
   // If non-null, use this instead of channels_->channel(0). This is an
   // optimization for the case num_proc_channels_ == 1 that allows us to point

webrtc/modules/audio_processing/level_estimator_impl.cc

@@ -12,6 +12,7 @@
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
@@ -29,13 +30,8 @@ int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
   }
 
   RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
-  if (audio->is_muted()) {
+  for (int i = 0; i < audio->num_channels(); ++i) {
-    rms_level->ProcessMuted(audio->samples_per_channel() *
+    rms_level->Process(audio->data(i), audio->samples_per_channel());
-                            audio->num_channels());
-  } else {
-    for (int i = 0; i < audio->num_channels(); ++i) {
-      rms_level->Process(audio->data(i), audio->samples_per_channel());
-    }
   }
 
   return AudioProcessing::kNoError;

webrtc/modules/audio_processing/rms_level.cc

@@ -15,16 +15,16 @@
 
 namespace webrtc {
 
-static const float kMaxSquaredLevel = 32768.0 * 32768.0;
+static const float kMaxSquaredLevel = 32768 * 32768;
 
 RMSLevel::RMSLevel()
-    : sum_square_(0.0),
+    : sum_square_(0),
       sample_count_(0) {}
 
 RMSLevel::~RMSLevel() {}
 
 void RMSLevel::Reset() {
-  sum_square_ = 0.0;
+  sum_square_ = 0;
   sample_count_ = 0;
 }
 
@@ -40,7 +40,7 @@ void RMSLevel::ProcessMuted(int length) {
 }
 
 int RMSLevel::RMS() {
-  if (sample_count_ == 0 || sum_square_ == 0.0) {
+  if (sample_count_ == 0 || sum_square_ == 0) {
     Reset();
     return kMinLevel;
   }

webrtc/modules/audio_processing/rms_level.h

@@ -8,6 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -49,3 +52,6 @@ class RMSLevel {
 };
 
 }  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+

webrtc/modules/audio_processing/test/audio_processing_unittest.cc

@@ -1229,15 +1229,6 @@ TEST_F(ApmTest, LevelEstimator) {
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
   EXPECT_EQ(70, apm_->level_estimator()->RMS());
 
-  // Min value if energy_ == 0.
-  SetFrameTo(frame_, 10000);
-  uint32_t energy = frame_->energy_;  // Save default to restore below.
-  frame_->energy_ = 0;
-  EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-  EXPECT_EQ(127, apm_->level_estimator()->RMS());
-  frame_->energy_ = energy;
-
   // Verify reset after enable/disable.
   SetFrameTo(frame_, 32767);
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));

webrtc/modules/utility/source/audio_frame_operations.cc

@@ -72,7 +72,6 @@ void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
 void AudioFrameOperations::Mute(AudioFrame& frame) {
   memset(frame.data_, 0, sizeof(int16_t) *
       frame.samples_per_channel_ * frame.num_channels_);
-  frame.energy_ = 0;
 }
 
 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {

webrtc/modules/utility/source/audio_frame_operations_unittest.cc

@@ -142,17 +142,13 @@ TEST_F(AudioFrameOperationsTest, SwapStereoChannelsFailsOnMono) {
 
 TEST_F(AudioFrameOperationsTest, MuteSucceeds) {
   SetFrameData(&frame_, 1000, 1000);
-  frame_.energy_ = 1000 * 1000 * frame_.samples_per_channel_ *
-      frame_.num_channels_;
   AudioFrameOperations::Mute(frame_);
 
   AudioFrame muted_frame;
   muted_frame.samples_per_channel_ = 320;
   muted_frame.num_channels_ = 2;
   SetFrameData(&muted_frame, 0, 0);
-  muted_frame.energy_ = 0;
   VerifyFramesAreEqual(muted_frame, frame_);
-  EXPECT_EQ(muted_frame.energy_, frame_.energy_);
 }
 
 // TODO(andrew): should not allow negative scales.

webrtc/voice_engine/channel.cc

@@ -3689,9 +3689,9 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
         MixOrReplaceAudioWithFile(mixingFrequency);
     }
 
-    if (Mute())
+    bool is_muted = Mute();  // Cache locally as Mute() takes a lock.
-    {
+    if (is_muted) {
-        AudioFrameOperations::Mute(_audioFrame);
+      AudioFrameOperations::Mute(_audioFrame);
     }
 
     if (channel_state_.Get().input_external_media)
@@ -3714,7 +3714,11 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
 
     if (_includeAudioLevelIndication) {
       int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
-      rms_level_.Process(_audioFrame.data_, length);
+      if (is_muted) {
+        rms_level_.ProcessMuted(length);
+      } else {
+        rms_level_.Process(_audioFrame.data_, length);
+      }
     }
 
     return 0;