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/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h
new file mode 100644
index 0000000..090c8fa
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/opus_test.h
@@ -0,0 +1,57 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+
+#include <math.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+
+namespace webrtc {
+
+class OpusTest : public ACMTest {
+ public:
+ OpusTest();
+ ~OpusTest();
+
+ void Perform();
+
+ private:
+ void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
+ int percent_loss = 0);
+
+ void OpenOutFile(int test_number);
+
+ rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
+ TestPackStereo* channel_a2b_;
+ PCMFile in_file_stereo_;
+ PCMFile in_file_mono_;
+ PCMFile out_file_;
+ PCMFile out_file_standalone_;
+ int counter_;
+ uint8_t payload_type_;
+ int rtp_timestamp_;
+ acm2::ACMResampler resampler_;
+ WebRtcOpusEncInst* opus_mono_encoder_;
+ WebRtcOpusEncInst* opus_stereo_encoder_;
+ WebRtcOpusDecInst* opus_mono_decoder_;
+ WebRtcOpusDecInst* opus_stereo_decoder_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_