fa58745445
They have all been replaced by AudioEncoder subclasses, accessed throgh ACMGenericCodecWrapper objects. After this change, the only subclass of ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated in a future cl.) This CL also deletes acm_opus_unittest.cc. This test file was already replaced audio_encoder_opus_unittest.cc in r8244. BUG=4228 COAUTHOR=kwiberg@webrtc.org R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40729004 Cr-Commit-Position: refs/heads/master@{#8457} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
57 lines
1.7 KiB
C++
57 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/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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#include "webrtc/system_wrappers/interface/scoped_ptr.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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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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