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84f75699c6
This change makes AudioCodingModule a pure sender and AcmReceiver a pure receiver. The Config struct is in practice no longer used by AudioCodingModule, so a new definition is included in AcmReceiver. The old definition remains in AudioCodingModule while downstream clients are being updated. Bug: webrtc:14867 Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533 Reviewed-by: Tomas Lundqvist <tomasl@google.com> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39244}
60 lines
1.6 KiB
C++
60 lines
1.6 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 MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
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#define MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
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#include <math.h>
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#include <memory>
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#include "modules/audio_coding/acm2/acm_receiver.h"
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#include "modules/audio_coding/acm2/acm_resampler.h"
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#include "modules/audio_coding/codecs/opus/opus_interface.h"
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#include "modules/audio_coding/test/PCMFile.h"
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#include "modules/audio_coding/test/TestStereo.h"
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namespace webrtc {
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class OpusTest {
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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,
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size_t channels,
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int bitrate,
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size_t frame_length,
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int percent_loss = 0);
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void OpenOutFile(int test_number);
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std::unique_ptr<acm2::AcmReceiver> 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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uint32_t 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 // MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
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