3c652b6746
Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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 R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
58 lines
1.7 KiB
C++
58 lines
1.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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#include <math.h>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
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#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
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#include "webrtc/modules/audio_coding/main/test/Channel.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
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namespace webrtc {
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class OpusTest : public ACMTest {
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public:
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OpusTest();
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~OpusTest();
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void Perform();
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private:
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void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
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int percent_loss = 0);
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void OpenOutFile(int test_number);
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rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
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TestPackStereo* channel_a2b_;
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PCMFile in_file_stereo_;
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PCMFile in_file_mono_;
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PCMFile out_file_;
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PCMFile out_file_standalone_;
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int counter_;
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uint8_t payload_type_;
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int rtp_timestamp_;
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acm2::ACMResampler resampler_;
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WebRtcOpusEncInst* opus_mono_encoder_;
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WebRtcOpusEncInst* opus_stereo_encoder_;
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WebRtcOpusDecInst* opus_mono_decoder_;
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WebRtcOpusDecInst* opus_stereo_decoder_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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