mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

This is a reland of 056f9738bf
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}
Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}
59 lines
1.6 KiB
C++
59 lines
1.6 KiB
C++
/*
|
|
* 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 MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
|
|
#define MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
|
|
|
|
#include <math.h>
|
|
|
|
#include <memory>
|
|
|
|
#include "modules/audio_coding/acm2/acm_resampler.h"
|
|
#include "modules/audio_coding/codecs/opus/opus_interface.h"
|
|
#include "modules/audio_coding/test/PCMFile.h"
|
|
#include "modules/audio_coding/test/TestStereo.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class OpusTest {
|
|
public:
|
|
OpusTest();
|
|
~OpusTest();
|
|
|
|
void Perform();
|
|
|
|
private:
|
|
void Run(TestPackStereo* channel,
|
|
size_t channels,
|
|
int bitrate,
|
|
size_t frame_length,
|
|
int percent_loss = 0);
|
|
|
|
void OpenOutFile(int test_number);
|
|
|
|
std::unique_ptr<AudioCodingModule> 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_;
|
|
uint32_t rtp_timestamp_;
|
|
acm2::ACMResampler resampler_;
|
|
WebRtcOpusEncInst* opus_mono_encoder_;
|
|
WebRtcOpusEncInst* opus_stereo_encoder_;
|
|
WebRtcOpusDecInst* opus_mono_decoder_;
|
|
WebRtcOpusDecInst* opus_stereo_decoder_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
|