audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
new file mode 100644
index 0000000..e38cd94
--- /dev/null
+++ b/webrtc/modules/audio_coding/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/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