Index: webrtc/modules/audio_coding/main/test/PacketLossTest.h |
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.h b/webrtc/modules/audio_coding/main/test/PacketLossTest.h |
deleted file mode 100644 |
index d25dea264f182e5ec335eb0888b3abb7f44af587..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/test/PacketLossTest.h |
+++ /dev/null |
@@ -1,67 +0,0 @@ |
-/* |
- * Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_ |
- |
-#include <string> |
-#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h" |
- |
-namespace webrtc { |
- |
-class ReceiverWithPacketLoss : public Receiver { |
- public: |
- ReceiverWithPacketLoss(); |
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream, |
- std::string out_file_name, int channels, int loss_rate, |
- int burst_length); |
- bool IncomingPacket() override; |
- |
- protected: |
- bool PacketLost(); |
- int loss_rate_; |
- int burst_length_; |
- int packet_counter_; |
- int lost_packet_counter_; |
- int burst_lost_counter_; |
-}; |
- |
-class SenderWithFEC : public Sender { |
- public: |
- SenderWithFEC(); |
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream, |
- std::string in_file_name, int sample_rate, int channels, |
- int expected_loss_rate); |
- bool SetPacketLossRate(int expected_loss_rate); |
- bool SetFEC(bool enable_fec); |
- protected: |
- int expected_loss_rate_; |
-}; |
- |
-class PacketLossTest : public ACMTest { |
- public: |
- PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate, |
- int burst_length); |
- void Perform(); |
- protected: |
- int channels_; |
- std::string in_file_name_; |
- int sample_rate_hz_; |
- rtc::scoped_ptr<SenderWithFEC> sender_; |
- rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_; |
- int expected_loss_rate_; |
- int actual_loss_rate_; |
- int burst_length_; |
-}; |
- |
-} // namespace webrtc |
- |
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_ |