/* * 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_MAIN_TEST_OPUS_TEST_H_ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_ #include #include "webrtc/modules/audio_coding/main/acm2/acm_opus.h" #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" #include "webrtc/modules/audio_coding/main/test/ACMTest.h" #include "webrtc/modules/audio_coding/main/test/Channel.h" #include "webrtc/modules/audio_coding/main/test/PCMFile.h" #include "webrtc/modules/audio_coding/main/test/TestStereo.h" #include "webrtc/system_wrappers/interface/scoped_ptr.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); scoped_ptr 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_MAIN_TEST_OPUS_TEST_H_