[MERGE TO 61] Fix incorrect memset on muted frames.

webrtc/modules/audio_coding/acm2/audio_coding_module.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/audio_coding/include/audio_coding_module.h"
 
+#include <algorithm>
+
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/safe_conversions.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
@@ skipping 305 matching lines @@
   RTC_DCHECK_GE(length_out_buff, frame.samples_per_channel_);
 
   if (!frame.muted()) {
     const int16_t* frame_data = frame.data();
     for (size_t n = 0; n < frame.samples_per_channel_; ++n) {
       out_buff[n] = static_cast<int16_t>(
           (static_cast<int32_t>(frame_data[2 * n]) +
            static_cast<int32_t>(frame_data[2 * n + 1])) >> 1);
     }
   } else {
-    memset(out_buff, 0, frame.samples_per_channel_);
+    std::fill(out_buff, out_buff + frame.samples_per_channel_, 0);
   }
   return 0;
 }
 
 // Mono-to-stereo can be used as in-place.
 int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
   RTC_DCHECK_EQ(frame.num_channels_, 1);
   RTC_DCHECK_GE(length_out_buff, 2 * frame.samples_per_channel_);
 
   if (!frame.muted()) {
     const int16_t* frame_data = frame.data();
     for (size_t n = frame.samples_per_channel_; n != 0; --n) {
       size_t i = n - 1;
       int16_t sample = frame_data[i];
       out_buff[2 * i + 1] = sample;
       out_buff[2 * i] = sample;
     }
   } else {
-    memset(out_buff, 0, 2 * frame.samples_per_channel_);
+    std::fill(out_buff, out_buff + frame.samples_per_channel_ * 2, 0);
   }
   return 0;
 }
 
 void ConvertEncodedInfoToFragmentationHeader(
     const AudioEncoder::EncodedInfo& info,
     RTPFragmentationHeader* frag) {
   if (info.redundant.empty()) {
     frag->fragmentationVectorSize = 0;
     return;
@@ skipping 990 matching lines @@
 
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     LOG(LS_ERROR) << "Invalid codec setting";
   return valid;
 }
 
 }  // namespace webrtc