1 files changed, 68 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
new file mode 100644
index 0000000000..cbcad85f5b
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
@@ -0,0 +1,68 @@
+/*
+ *  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/acm2/acm_resampler.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/system_wrappers/include/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,
+                                 int num_audio_channels,
+                                 size_t out_capacity_samples,
+                                 int16_t* out_audio) {
+  size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
+  int out_length = out_freq_hz * num_audio_channels / 100;
+  if (in_freq_hz == out_freq_hz) {
+    if (out_capacity_samples < in_length) {
+      assert(false);
+      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) {
+    LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
+              num_audio_channels);
+    return -1;
+  }
+
+  out_length =
+      resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
+  if (out_length == -1) {
+    LOG_FERR4(LS_ERROR,
+              Resample,
+              in_audio,
+              in_length,
+              out_audio,
+              out_capacity_samples);
+    return -1;
+  }
+
+  return out_length / num_audio_channels;
+}
+
+}  // namespace acm2
+}  // namespace webrtc