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diff --git a/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.cc b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.cc
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+++ b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.cc
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+/*
+ * Copyright (c) 2012 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.
+ */
+
+#include "TwoWayCommunication.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+TwoWayCommunication::TwoWayCommunication()
+ : _acmA(AudioCodingModule::Create(
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+ _acmRefA(AudioCodingModule::Create(
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {
+ AudioCodingModule::Config config;
+ // The clicks will be more obvious if time-stretching is not allowed.
+ // TODO(henrik.lundin) Really?
+ config.neteq_config.for_test_no_time_stretching = true;
+ config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+ _acmB.reset(AudioCodingModule::Create(config));
+ _acmRefB.reset(AudioCodingModule::Create(config));
+}
+
+TwoWayCommunication::~TwoWayCommunication() {
+ delete _channel_A2B;
+ delete _channel_B2A;
+ delete _channelRef_A2B;
+ delete _channelRef_B2A;
+ _inFileA.Close();
+ _inFileB.Close();
+ _outFileA.Close();
+ _outFileB.Close();
+ _outFileRefA.Close();
+ _outFileRefB.Close();
+}
+
+void TwoWayCommunication::SetUpAutotest(
+ AudioEncoderFactory* const encoder_factory,
+ const SdpAudioFormat& format1,
+ const int payload_type1,
+ const SdpAudioFormat& format2,
+ const int payload_type2) {
+ //--- Set A codecs
+ _acmA->SetEncoder(
+ encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
+ _acmA->SetReceiveCodecs({{payload_type2, format2}});
+
+ //--- Set ref-A codecs
+ _acmRefA->SetEncoder(
+ encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
+ _acmRefA->SetReceiveCodecs({{payload_type2, format2}});
+
+ //--- Set B codecs
+ _acmB->SetEncoder(
+ encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
+ _acmB->SetReceiveCodecs({{payload_type1, format1}});
+
+ //--- Set ref-B codecs
+ _acmRefB->SetEncoder(
+ encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
+ _acmRefB->SetReceiveCodecs({{payload_type1, format1}});
+
+ uint16_t frequencyHz;
+
+ //--- Input A and B
+ std::string in_file_name =
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+ frequencyHz = 16000;
+ _inFileA.Open(in_file_name, frequencyHz, "rb");
+ _inFileB.Open(in_file_name, frequencyHz, "rb");
+
+ //--- Output A
+ std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm";
+ frequencyHz = 16000;
+ _outFileA.Open(output_file_a, frequencyHz, "wb");
+ std::string output_ref_file_a =
+ webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
+ _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb");
+
+ //--- Output B
+ std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm";
+ frequencyHz = 16000;
+ _outFileB.Open(output_file_b, frequencyHz, "wb");
+ std::string output_ref_file_b =
+ webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
+ _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb");
+
+ //--- Set A-to-B channel
+ _channel_A2B = new Channel;
+ _acmA->RegisterTransportCallback(_channel_A2B);
+ _channel_A2B->RegisterReceiverACM(_acmB.get());
+ //--- Do the same for the reference
+ _channelRef_A2B = new Channel;
+ _acmRefA->RegisterTransportCallback(_channelRef_A2B);
+ _channelRef_A2B->RegisterReceiverACM(_acmRefB.get());
+
+ //--- Set B-to-A channel
+ _channel_B2A = new Channel;
+ _acmB->RegisterTransportCallback(_channel_B2A);
+ _channel_B2A->RegisterReceiverACM(_acmA.get());
+ //--- Do the same for reference
+ _channelRef_B2A = new Channel;
+ _acmRefB->RegisterTransportCallback(_channelRef_B2A);
+ _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
+}
+
+void TwoWayCommunication::Perform() {
+ const SdpAudioFormat format1("ISAC", 16000, 1);
+ const SdpAudioFormat format2("L16", 8000, 1);
+ constexpr int payload_type1 = 17, payload_type2 = 18;
+
+ auto encoder_factory = CreateBuiltinAudioEncoderFactory();
+
+ SetUpAutotest(encoder_factory.get(), format1, payload_type1, format2,
+ payload_type2);
+
+ unsigned int msecPassed = 0;
+ unsigned int secPassed = 0;
+
+ int32_t outFreqHzA = _outFileA.SamplingFrequency();
+ int32_t outFreqHzB = _outFileB.SamplingFrequency();
+
+ AudioFrame audioFrame;
+
+ // In the following loop we tests that the code can handle misuse of the APIs.
+ // In the middle of a session with data flowing between two sides, called A
+ // and B, APIs will be called, and the code should continue to run, and be
+ // able to recover.
+ while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
+ msecPassed += 10;
+ EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+ EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+ EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0);
+
+ EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0);
+
+ EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
+ EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
+ bool muted;
+ EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
+ _outFileA.Write10MsData(audioFrame);
+ EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
+ _outFileRefA.Write10MsData(audioFrame);
+ EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
+ _outFileB.Write10MsData(audioFrame);
+ EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
+ _outFileRefB.Write10MsData(audioFrame);
+
+ // Update time counters each time a second of data has passed.
+ if (msecPassed >= 1000) {
+ msecPassed = 0;
+ secPassed++;
+ }
+ // Re-register send codec on side B.
+ if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
+ _acmB->SetEncoder(encoder_factory->MakeAudioEncoder(
+ payload_type2, format2, absl::nullopt));
+ }
+ // Initialize receiver on side A.
+ if (((secPassed % 7) == 6) && (msecPassed == 0))
+ EXPECT_EQ(0, _acmA->InitializeReceiver());
+ // Re-register codec on side A.
+ if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
+ _acmA->SetReceiveCodecs({{payload_type2, format2}});
+ }
+ }
+}
+
+} // namespace webrtc