diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
index 437feb7595..8ec8231112 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -123,7 +123,10 @@ int32_t ACMGenericCodec::Add10MsDataSafe(const uint32_t timestamp,
     if ((in_audio_ix_write_ >= length_smpl * audio_channel) &&
         (in_timestamp_ix_write_ > 0)) {
       in_audio_ix_write_ -= length_smpl * audio_channel;
+      assert(in_timestamp_ix_write_ >= 0);
+
       in_timestamp_ix_write_--;
+      assert(in_audio_ix_write_ >= 0);
       WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, unique_id_,
                    "Adding 10ms with previous timestamp, overwriting the "
                    "previous 10ms");
@@ -160,8 +163,11 @@ int32_t ACMGenericCodec::Add10MsDataSafe(const uint32_t timestamp,
     memmove(in_timestamp_, in_timestamp_ + missed_10ms_blocks,
             (in_timestamp_ix_write_ - missed_10ms_blocks) * sizeof(uint32_t));
     in_timestamp_ix_write_ -= missed_10ms_blocks;
+    assert(in_timestamp_ix_write_ >= 0);
+
     in_timestamp_[in_timestamp_ix_write_] = timestamp;
     in_timestamp_ix_write_++;
+    assert(in_timestamp_ix_write_ < TIMESTAMP_BUFFER_SIZE_W32);
 
     // Buffer is full.
     in_audio_ix_write_ = AUDIO_BUFFER_SIZE_W16;
@@ -173,12 +179,11 @@ int32_t ACMGenericCodec::Add10MsDataSafe(const uint32_t timestamp,
   memcpy(in_audio_ + in_audio_ix_write_, data,
          length_smpl * audio_channel * sizeof(int16_t));
   in_audio_ix_write_ += length_smpl * audio_channel;
-
   assert(in_timestamp_ix_write_ < TIMESTAMP_BUFFER_SIZE_W32);
-  assert(in_timestamp_ix_write_ >= 0);
 
   in_timestamp_[in_timestamp_ix_write_] = timestamp;
   in_timestamp_ix_write_++;
+  assert(in_timestamp_ix_write_ < TIMESTAMP_BUFFER_SIZE_W32);
   return 0;
 }
 
@@ -345,6 +350,7 @@ int16_t ACMGenericCodec::Encode(uint8_t* bitstream,
             (in_timestamp_ix_write_ - num_10ms_blocks) * sizeof(int32_t));
   }
   in_timestamp_ix_write_ -= num_10ms_blocks;
+  assert(in_timestamp_ix_write_ >= 0);
 
   // Remove encoded audio and move next audio to be encoded to the beginning
   // of the buffer. Accordingly, adjust the read and write indices.
@@ -448,11 +454,8 @@ int16_t ACMGenericCodec::InitEncoderSafe(WebRtcACMCodecParams* codec_params,
   int mirrorID;
   int codec_number = ACMCodecDB::CodecNumber(codec_params->codec_inst,
                                              &mirrorID);
-  if (codec_number < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
-                 "InitEncoderSafe: error, codec number negative");
-    return -1;
-  }
+  assert(codec_number >= 0);
+
   // Check if the parameters are for this codec.
   if ((codec_id_ >= 0) && (codec_id_ != codec_number) &&
       (codec_id_ != mirrorID)) {
@@ -481,7 +484,7 @@ int16_t ACMGenericCodec::InitEncoderSafe(WebRtcACMCodecParams* codec_params,
       encoder_exist_ = true;
     }
   }
-  frame_len_smpl_ = (codec_params->codec_inst).pacsize;
+  frame_len_smpl_ = codec_params->codec_inst.pacsize;
   num_channels_ = codec_params->codec_inst.channels;
   status = InternalInitEncoder(codec_params);
   if (status < 0) {
@@ -490,24 +493,25 @@ int16_t ACMGenericCodec::InitEncoderSafe(WebRtcACMCodecParams* codec_params,
     encoder_initialized_ = false;
     return -1;
   } else {
+    // TODO(turajs): Move these allocations to the constructor issue 2445.
     // Store encoder parameters.
     memcpy(&encoder_params_, codec_params, sizeof(WebRtcACMCodecParams));
     encoder_initialized_ = true;
     if (in_audio_ == NULL) {
       in_audio_ = new int16_t[AUDIO_BUFFER_SIZE_W16];
-      if (in_audio_ == NULL) {
-        return -1;
-      }
-      memset(in_audio_, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t));
     }
     if (in_timestamp_ == NULL) {
       in_timestamp_ = new uint32_t[TIMESTAMP_BUFFER_SIZE_W32];
-      if (in_timestamp_ == NULL) {
-        return -1;
-      }
-      memset(in_timestamp_, 0, sizeof(uint32_t) * TIMESTAMP_BUFFER_SIZE_W32);
     }
   }
+
+  // Fresh start of audio buffer.
+  memset(in_audio_, 0, sizeof(*in_audio_) * AUDIO_BUFFER_SIZE_W16);
+  memset(in_timestamp_, 0, sizeof(*in_timestamp_) * TIMESTAMP_BUFFER_SIZE_W32);
+  in_audio_ix_write_ = 0;
+  in_audio_ix_read_ = 0;
+  in_timestamp_ix_write_ = 0;
+
   return SetVADSafe(&codec_params->enable_dtx, &codec_params->enable_vad,
                     &codec_params->vad_mode);
 }
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
index fe0a13ca4c..4e53b873a1 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
@@ -170,6 +170,7 @@ int32_t ACMGenericCodec::Add10MsDataSafe(const uint32_t timestamp,
     memmove(in_timestamp_, in_timestamp_ + missed_10ms_blocks,
             (in_timestamp_ix_write_ - missed_10ms_blocks) * sizeof(uint32_t));
     in_timestamp_ix_write_ -= missed_10ms_blocks;
+    assert(in_timestamp_ix_write_ >= 0);
     in_timestamp_[in_timestamp_ix_write_] = timestamp;
     in_timestamp_ix_write_++;
 
@@ -351,7 +352,7 @@ int16_t ACMGenericCodec::Encode(uint8_t* bitstream,
             (in_timestamp_ix_write_ - num_10ms_blocks) * sizeof(int32_t));
   }
   in_timestamp_ix_write_ -= num_10ms_blocks;
-
+  assert(in_timestamp_ix_write_ >= 0);
   // Remove encoded audio and move next audio to be encoded to the beginning
   // of the buffer. Accordingly, adjust the read and write indices.
   if (in_audio_ix_read_ < in_audio_ix_write_) {
@@ -359,6 +360,7 @@ int16_t ACMGenericCodec::Encode(uint8_t* bitstream,
             (in_audio_ix_write_ - in_audio_ix_read_) * sizeof(int16_t));
   }
   in_audio_ix_write_ -= in_audio_ix_read_;
+  assert(in_timestamp_ix_write_ >= 0);
   in_audio_ix_read_ = 0;
   last_encoded_timestamp_ = *timestamp;
   return (status < 0) ? (-1) : (*bitstream_len_byte);
@@ -574,20 +576,23 @@ int16_t ACMGenericCodec::InitEncoderSafe(WebRtcACMCodecParams* codec_params,
       if (in_audio_ == NULL) {
         return -1;
       }
-      memset(in_audio_, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t));
     }
     if (in_timestamp_ == NULL) {
       in_timestamp_ = new uint32_t[TIMESTAMP_BUFFER_SIZE_W32];
       if (in_timestamp_ == NULL) {
         return -1;
       }
-      memset(in_timestamp_, 0, sizeof(uint32_t) * TIMESTAMP_BUFFER_SIZE_W32);
     }
+    // Fresh start for audio buffer.
     is_audio_buff_fresh_ = true;
+    memset(in_audio_, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(int16_t));
+    memset(in_timestamp_, 0, sizeof(uint32_t) * TIMESTAMP_BUFFER_SIZE_W32);
+    in_audio_ix_write_ = 0;
+    in_audio_ix_read_ = 0;
+    in_timestamp_ix_write_ = 0;
   }
   status = SetVADSafe(&codec_params->enable_dtx, &codec_params->enable_vad,
                       &codec_params->vad_mode);
-
   return status;
 }