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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
77 lines
2 KiB
C++
77 lines
2 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 : public ACMTest {
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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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