Index: webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
deleted file mode 100644 |
index cbcad85f5bc7b3447e08493f172cb15b21060427..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc |
+++ /dev/null |
@@ -1,68 +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/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 |