Avoid clicks when muting/unmuting a voe::Channel.

webrtc/modules/utility/source/audio_frame_operations.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/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
+namespace  {
+
+// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
+const size_t kMuteFadeFrames = 128;
+const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
+
+}  // namespace {
 
 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
                                         size_t samples_per_channel,
                                         int16_t* dst_audio) {
   for (size_t i = 0; i < samples_per_channel; i++) {
     dst_audio[2 * i] = src_audio[i];
     dst_audio[2 * i + 1] = src_audio[i];
   }
 }
 
@@ skipping 37 matching lines @@
 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
   if (frame->num_channels_ != 2) return;
 
   for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     int16_t temp_data = frame->data_[i];
     frame->data_[i] = frame->data_[i + 1];
     frame->data_[i + 1] = temp_data;
   }
 }
 
-void AudioFrameOperations::Mute(AudioFrame& frame) {
-  memset(frame.data_, 0, sizeof(int16_t) *
-      frame.samples_per_channel_ * frame.num_channels_);
+void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
+                                bool current_frame_muted) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK(frame->interleaved_);
+  if (!previous_frame_muted && !current_frame_muted) {
+    // Not muted, don't touch.
+  } else if (previous_frame_muted && current_frame_muted) {
+    // Frame fully muted.
+    size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
+    RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
+    memset(frame->data_, 0, sizeof(frame->data_[0]) * total_samples);
+  } else {
+    // Limit number of samples to fade, if frame isn't long enough.
+    size_t count = kMuteFadeFrames;
+    float inc = kMuteFadeInc;
+    if (frame->samples_per_channel_ < kMuteFadeFrames) {
+      count = frame->samples_per_channel_;
+      if (count > 0) {
+        inc = 1.0f / count;
+      }
+    }
+
+    size_t start = 0;
+    float g = 0.0f;
+    if (current_frame_muted) {
+      // Fade out the last |count| samples of frame.
+      RTC_DCHECK(!previous_frame_muted);
+      start = frame->samples_per_channel_ - count;
+      g = 1.0f;
+      inc = -inc;
+    } else {
+      // Fade in the first |count| samples of frame.
+      RTC_DCHECK(previous_frame_muted);
+    }
+
+    // Perform fade.
+    size_t end = (start + count);
+    size_t channels = frame->num_channels_;
+    for (size_t i = start * channels; i < end * channels; i += channels) {
+      g += inc;
+      for (size_t j = 0; j < channels; ++j) {
+        frame->data_[i + j] *= g;
+      }
+    }
+  }
 }
 
 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
   if (frame.num_channels_ != 2) {
     return -1;
   }
 
   for (size_t i = 0; i < frame.samples_per_channel_; i++) {
     frame.data_[2 * i] =
         static_cast<int16_t>(left * frame.data_[2 * i]);
@@ skipping 15 matching lines @@
     } else if (temp_data > 32767) {
       frame.data_[i] = 32767;
     } else {
       frame.data_[i] = static_cast<int16_t>(temp_data);
     }
   }
   return 0;
 }
 
 }  // namespace webrtc