1 files changed, 0 insertions, 63 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc
deleted file mode 100644
index 50ddab1d8b9..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc
+++ /dev/null
@@ -1,63 +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.
- */
-
-#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
-
-#include <string.h>
-
-#include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-namespace webrtc {
-
-namespace acm1 {
-
-ACMResampler::ACMResampler() {
-}
-
-ACMResampler::~ACMResampler() {
-}
-
-int16_t ACMResampler::Resample10Msec(const int16_t* in_audio,
-                                     int32_t in_freq_hz,
-                                     int16_t* out_audio,
-                                     int32_t out_freq_hz,
-                                     uint8_t num_audio_channels) {
-  if (in_freq_hz == out_freq_hz) {
-    size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
-    memcpy(out_audio, in_audio, length * sizeof(int16_t));
-    return static_cast<int16_t>(in_freq_hz / 100);
-  }
-
-  // |max_length| is the maximum number of samples for 10ms at 48kHz.
-  // TODO(turajs): is this actually the capacity of the |out_audio| buffer?
-  int max_length = 480 * num_audio_channels;
-  int in_length = in_freq_hz / 100 * num_audio_channels;
-
-  if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
-                                    num_audio_channels) != 0) {
-    LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
-              num_audio_channels);
-    return -1;
-  }
-
-  int out_length = resampler_.Resample(in_audio, in_length, out_audio,
-                                       max_length);
-  if (out_length == -1) {
-    LOG_FERR4(LS_ERROR, Resample, in_audio, in_length, out_audio, max_length);
-    return -1;
-  }
-
-  return out_length / num_audio_channels;
-}
-
-}  // namespace acm1
-
-}  // namespace webrtc