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diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.h b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.h
deleted file mode 100644
index b50e722c443..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
-
-#include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-namespace acm1 {
-
-class ACMResampler {
- public:
- ACMResampler();
- ~ACMResampler();
-
- int16_t Resample10Msec(const int16_t* in_audio,
- const int32_t in_freq_hz,
- int16_t* out_audio,
- const int32_t out_freq_hz,
- uint8_t num_audio_channels);
-
- private:
- PushResampler resampler_;
-};
-
-} // namespace acm1
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_