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ec0f45be11
This reverts commit 056f9738bf.
Reason for revert: breaks downstream
Original change's description:
> Remove CodecInst pt.1
>
> Update audio_coding tests to not use CodecInst.
>
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}
TBR=solenberg@webrtc.org,kwiberg@webrtc.org
Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}
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_resampler.h"
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#include "modules/audio_coding/codecs/opus/opus_interface.h"
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#include "modules/audio_coding/test/Channel.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<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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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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