1 files changed, 61 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc
new file mode 100644
index 0000000000..e307c6ca57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc
@@ -0,0 +1,61 @@
+/*
+ *  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 "modules/audio_coding/acm2/acm_resampler.h"
+
+#include <string.h>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace acm2 {
+
+ACMResampler::ACMResampler() {}
+
+ACMResampler::~ACMResampler() {}
+
+int ACMResampler::Resample10Msec(const int16_t* in_audio,
+                                 int in_freq_hz,
+                                 int out_freq_hz,
+                                 size_t num_audio_channels,
+                                 size_t out_capacity_samples,
+                                 int16_t* out_audio) {
+  size_t in_length = in_freq_hz * num_audio_channels / 100;
+  if (in_freq_hz == out_freq_hz) {
+    if (out_capacity_samples < in_length) {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+    memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
+    return static_cast<int>(in_length / num_audio_channels);
+  }
+
+  if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
+                                    num_audio_channels) != 0) {
+    RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
+                      << out_freq_hz << ", " << num_audio_channels
+                      << ") failed.";
+    return -1;
+  }
+
+  int out_length =
+      resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
+  if (out_length == -1) {
+    RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
+                      << out_audio << ", " << out_capacity_samples
+                      << ") failed.";
+    return -1;
+  }
+
+  return static_cast<int>(out_length / num_audio_channels);
+}
+
+}  // namespace acm2
+}  // namespace webrtc