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Unified Diff: webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
deleted file mode 100644
index 725cbf74d720fd40bab5b02c7e3b0d57f1821241..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ /dev/null
@@ -1,299 +0,0 @@
-/*
- * 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 <ctype.h>
-#include <stdio.h>
-#include <string.h>
-
-#ifdef WIN32
-#include <Windows.h>
-#endif
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-TwoWayCommunication::TwoWayCommunication(int testMode)
- : _acmA(AudioCodingModule::Create(1)),
- _acmRefA(AudioCodingModule::Create(3)),
- _testMode(testMode) {
- AudioCodingModule::Config config;
- // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
- config.neteq_config.playout_mode = kPlayoutFax;
- config.id = 2;
- _acmB.reset(AudioCodingModule::Create(config));
- config.id = 4;
- _acmRefB.reset(AudioCodingModule::Create(config));
-}
-
-TwoWayCommunication::~TwoWayCommunication() {
- delete _channel_A2B;
- delete _channel_B2A;
- delete _channelRef_A2B;
- delete _channelRef_B2A;
-#ifdef WEBRTC_DTMF_DETECTION
- if (_dtmfDetectorA != NULL) {
- delete _dtmfDetectorA;
- }
- if (_dtmfDetectorB != NULL) {
- delete _dtmfDetectorB;
- }
-#endif
- _inFileA.Close();
- _inFileB.Close();
- _outFileA.Close();
- _outFileB.Close();
- _outFileRefA.Close();
- _outFileRefB.Close();
-}
-
-void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
- uint8_t* codecID_B) {
- rtc::scoped_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
- uint8_t noCodec = tmpACM->NumberOfCodecs();
- CodecInst codecInst;
- printf("List of Supported Codecs\n");
- printf("========================\n");
- for (uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++) {
- EXPECT_EQ(tmpACM->Codec(codecCntr, &codecInst), 0);
- printf("%d- %s\n", codecCntr, codecInst.plname);
- }
- printf("\nChoose a send codec for side A [0]: ");
- char myStr[15] = "";
- EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
- *codecID_A = (uint8_t) atoi(myStr);
-
- printf("\nChoose a send codec for side B [0]: ");
- EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
- *codecID_B = (uint8_t) atoi(myStr);
-
- printf("\n");
-}
-
-void TwoWayCommunication::SetUp() {
- uint8_t codecID_A;
- uint8_t codecID_B;
-
- ChooseCodec(&codecID_A, &codecID_B);
- CodecInst codecInst_A;
- CodecInst codecInst_B;
- CodecInst dummyCodec;
- EXPECT_EQ(0, _acmA->Codec(codecID_A, &codecInst_A));
- EXPECT_EQ(0, _acmB->Codec(codecID_B, &codecInst_B));
- EXPECT_EQ(0, _acmA->Codec(6, &dummyCodec));
-
- //--- Set A codecs
- EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
- EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
- //--- Set ref-A codecs
- EXPECT_EQ(0, _acmRefA->RegisterSendCodec(codecInst_A));
- EXPECT_EQ(0, _acmRefA->RegisterReceiveCodec(codecInst_B));
-
- //--- Set B codecs
- EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
- EXPECT_EQ(0, _acmB->RegisterReceiveCodec(codecInst_A));
-
- //--- Set ref-B codecs
- EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
- EXPECT_EQ(0, _acmRefB->RegisterReceiveCodec(codecInst_A));
-
- uint16_t frequencyHz;
-
- //--- Input A
- std::string in_file_name = webrtc::test::ResourcePath(
- "audio_coding/testfile32kHz", "pcm");
- frequencyHz = 32000;
- printf("Enter input file at side A [%s]: ", in_file_name.c_str());
- PCMFile::ChooseFile(&in_file_name, 499, &frequencyHz);
- _inFileA.Open(in_file_name, frequencyHz, "rb");
-
- //--- Output A
- std::string out_file_a = webrtc::test::OutputPath() + "outA.pcm";
- printf("Output file at side A: %s\n", out_file_a.c_str());
- printf("Sampling frequency (in Hz) of the above file: %u\n", frequencyHz);
- _outFileA.Open(out_file_a, frequencyHz, "wb");
- std::string ref_file_name = webrtc::test::OutputPath() + "ref_outA.pcm";
- _outFileRefA.Open(ref_file_name, frequencyHz, "wb");
-
- //--- Input B
- in_file_name = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
- "pcm");
- frequencyHz = 32000;
- printf("\n\nEnter input file at side B [%s]: ", in_file_name.c_str());
- PCMFile::ChooseFile(&in_file_name, 499, &frequencyHz);
- _inFileB.Open(in_file_name, frequencyHz, "rb");
-
- //--- Output B
- std::string out_file_b = webrtc::test::OutputPath() + "outB.pcm";
- printf("Output file at side B: %s\n", out_file_b.c_str());
- printf("Sampling frequency (in Hz) of the above file: %u\n", frequencyHz);
- _outFileB.Open(out_file_b, frequencyHz, "wb");
- ref_file_name = webrtc::test::OutputPath() + "ref_outB.pcm";
- _outFileRefB.Open(ref_file_name, 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::SetUpAutotest() {
- CodecInst codecInst_A;
- CodecInst codecInst_B;
- CodecInst dummyCodec;
-
- EXPECT_EQ(0, _acmA->Codec("ISAC", &codecInst_A, 16000, 1));
- EXPECT_EQ(0, _acmB->Codec("L16", &codecInst_B, 8000, 1));
- EXPECT_EQ(0, _acmA->Codec(6, &dummyCodec));
-
- //--- Set A codecs
- EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
- EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
-
- //--- Set ref-A codecs
- EXPECT_GT(_acmRefA->RegisterSendCodec(codecInst_A), -1);
- EXPECT_GT(_acmRefA->RegisterReceiveCodec(codecInst_B), -1);
-
- //--- Set B codecs
- EXPECT_GT(_acmB->RegisterSendCodec(codecInst_B), -1);
- EXPECT_GT(_acmB->RegisterReceiveCodec(codecInst_A), -1);
-
- //--- Set ref-B codecs
- EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
- EXPECT_EQ(0, _acmRefB->RegisterReceiveCodec(codecInst_A));
-
- 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() {
- if (_testMode == 0) {
- SetUpAutotest();
- } else {
- SetUp();
- }
- unsigned int msecPassed = 0;
- unsigned int secPassed = 0;
-
- int32_t outFreqHzA = _outFileA.SamplingFrequency();
- int32_t outFreqHzB = _outFileB.SamplingFrequency();
-
- AudioFrame audioFrame;
-
- auto codecInst_B = _acmB->SendCodec();
- ASSERT_TRUE(codecInst_B);
-
- // 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);
- EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame));
- _outFileA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame));
- _outFileRefA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
- _outFileB.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame));
- _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)) {
- EXPECT_EQ(0, _acmB->RegisterSendCodec(*codecInst_B));
- EXPECT_TRUE(_acmB->SendCodec());
- }
- // 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)) {
- EXPECT_EQ(0, _acmA->RegisterReceiveCodec(*codecInst_B));
- }
- }
-}
-
-} // namespace webrtc
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