audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.h b/webrtc/modules/audio_coding/test/PacketLossTest.h
new file mode 100644
index 0000000..f3570ae
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/PacketLossTest.h
@@ -0,0 +1,67 @@
+/*
+ * 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_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+
+#include <string>
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/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_TEST_PACKETLOSSTEST_H_