ec0f45be11
This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851. 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}
78 lines
2.0 KiB
C++
78 lines
2.0 KiB
C++
/*
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* Copyright (c) 2014 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_PACKETLOSSTEST_H_
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#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
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#include <memory>
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#include <string>
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#include "modules/audio_coding/test/EncodeDecodeTest.h"
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namespace webrtc {
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class ReceiverWithPacketLoss : public Receiver {
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public:
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ReceiverWithPacketLoss();
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void Setup(AudioCodingModule* acm,
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RTPStream* rtpStream,
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std::string out_file_name,
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int channels,
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int loss_rate,
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int burst_length);
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bool IncomingPacket() override;
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protected:
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bool PacketLost();
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int loss_rate_;
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int burst_length_;
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int packet_counter_;
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int lost_packet_counter_;
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int burst_lost_counter_;
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};
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class SenderWithFEC : public Sender {
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public:
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SenderWithFEC();
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void Setup(AudioCodingModule* acm,
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RTPStream* rtpStream,
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std::string in_file_name,
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int sample_rate,
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int channels,
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int expected_loss_rate);
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bool SetPacketLossRate(int expected_loss_rate);
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bool SetFEC(bool enable_fec);
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protected:
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int expected_loss_rate_;
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};
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class PacketLossTest {
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public:
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PacketLossTest(int channels,
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int expected_loss_rate_,
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int actual_loss_rate,
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int burst_length);
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void Perform();
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protected:
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int channels_;
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std::string in_file_name_;
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int sample_rate_hz_;
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std::unique_ptr<SenderWithFEC> sender_;
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std::unique_ptr<ReceiverWithPacketLoss> receiver_;
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int expected_loss_rate_;
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int actual_loss_rate_;
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int burst_length_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
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