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657b296ff5
This is a reland of 056f9738bf
Original change's description:
> Remove CodecInst pt.1
>
> Update audio_coding tests to not use CodecInst.
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> 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}
75 lines
1.9 KiB
C++
75 lines
1.9 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 <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 file_num,
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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 payload_type,
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SdpAudioFormat format,
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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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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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