Adding a recording tool to vie autottest
Review URL: https://webrtc-codereview.appspot.com/746006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2669 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mikhal@webrtc.org
parent
36d0baf208
commit
36b95b4753
@ -59,6 +59,7 @@ public:
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int ViELoopbackCall();
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int ViESimulcastCall();
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int ViECustomCall();
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int ViERecordCall();
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// All functions except the three above are meant to run in a
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// googletest harness.
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@ -117,7 +117,7 @@ int ViEAutoTestMain::RunSpecificTestCaseIn(const std::string test_case_name)
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int ViEAutoTestMain::RunSpecialTestCase(int choice) {
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// 7-9 don't run in GTest and need to initialize by themselves.
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assert(choice >= 7 && choice <= 9);
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assert(choice >= 7 && choice <= 10);
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// Create the windows
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ViEWindowCreator windowCreator;
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@ -133,6 +133,7 @@ int ViEAutoTestMain::RunSpecialTestCase(int choice) {
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case 7: errors = vieAutoTest.ViELoopbackCall(); break;
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case 8: errors = vieAutoTest.ViECustomCall(); break;
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case 9: errors = vieAutoTest.ViESimulcastCall(); break;
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case 10: errors = vieAutoTest.ViERecordCall(); break;
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}
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windowCreator.TerminateWindows();
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@ -158,9 +159,10 @@ int ViEAutoTestMain::RunInteractiveMode() {
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ViETest::Log("\t 7. Simple loopback call");
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ViETest::Log("\t 8. Custom configure a call");
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ViETest::Log("\t 9. Simulcast in loopback");
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ViETest::Log("\t 10. Record");
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ViETest::Log("Select type of test:");
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choice = AskUserForNumber(0, 9);
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choice = AskUserForNumber(0, 10);
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if (choice == kInvalidChoice) {
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continue;
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}
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588
src/video_engine/test/auto_test/source/vie_autotest_record.cc
Normal file
588
src/video_engine/test/auto_test/source/vie_autotest_record.cc
Normal file
@ -0,0 +1,588 @@
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/*
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* Copyright (c) 2012 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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//
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// vie_autotest_record.cc
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//
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// This code is also used as sample code for ViE 3.0
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//
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#include <stdio.h>
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#include <fstream>
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#include "common_types.h"
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#include "video_engine/test/libvietest/include/tb_external_transport.h"
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#include "voice_engine/include/voe_base.h"
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#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
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#include "video_engine/test/auto_test/interface/vie_autotest.h"
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#include "video_engine/include/vie_base.h"
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#include "video_engine/include/vie_capture.h"
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#include "video_engine/include/vie_codec.h"
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#include "video_engine/include/vie_network.h"
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#include "video_engine/include/vie_render.h"
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#include "video_engine/include/vie_rtp_rtcp.h"
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#include "voice_engine/include/voe_rtp_rtcp.h"
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#include "system_wrappers/interface/tick_util.h"
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#define VCM_RED_PAYLOAD_TYPE 96
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#define VCM_ULPFEC_PAYLOAD_TYPE 97
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#define DEFAULT_AUDIO_PORT 11113
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#define DEFAULT_AUDIO_CODEC "ISAC"
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#define DEFAULT_VIDEO_CODEC_WIDTH 640
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#define DEFAULT_VIDEO_CODEC_HEIGHT 480
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#define DEFAULT_VIDEO_CODEC_START_RATE 1000
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#define DEFAULT_RECORDING_FOLDER "RECORDING"
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#define DEFAULT_RECORDING_AUDIO "/audio_debug.aec"
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#define DEFAULT_RECORDING_AUDIO_RTP "/audio_rtpdump.rtp"
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#define DEFAULT_RECORDING_VIDEO_RTP "/video_rtpdump.rtp"
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bool GetAudioDevices(webrtc::VoEBase* voe_base,
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webrtc::VoEHardware* voe_hardware,
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char* recording_device_name,
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int& recording_device_index,
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char* playbackDeviceName,
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int& playback_device_index);
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bool GetAudioCodecRecord(webrtc::VoECodec* voe_codec,
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webrtc::CodecInst& audio_codec);
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int VideoEngineSampleRecordCode(void* window1, void* window2) {
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int error = 0;
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// Audio settings.
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int audio_tx_port = DEFAULT_AUDIO_PORT;
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int audio_rx_port = DEFAULT_AUDIO_PORT;
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webrtc::CodecInst audio_codec;
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int audio_channel = -1;
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int audio_capture_device_index = -1;
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int audio_playback_device_index = -1;
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const unsigned int KMaxDeviceNameLength = 128;
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const unsigned int KMaxUniqueIdLength = 256;
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char deviceName[KMaxDeviceNameLength];
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char audio_capture_device_name[KMaxUniqueIdLength] = "";
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char audio_playbackDeviceName[KMaxUniqueIdLength] = "";
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// Network settings.
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const char* ipAddress = "127.0.0.1";
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const int rtpPort = 6000;
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//
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// Create a VideoEngine instance
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//
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webrtc::VideoEngine* ptrViE = NULL;
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ptrViE = webrtc::VideoEngine::Create();
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if (ptrViE == NULL) {
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printf("ERROR in VideoEngine::Create\n");
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return -1;
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}
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error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
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if (error == -1) {
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printf("ERROR in VideoEngine::SetTraceLevel\n");
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return -1;
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}
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std::string trace_file =
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ViETest::GetResultOutputPath() + "ViERecordCall_trace.txt";
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error = ptrViE->SetTraceFile(trace_file.c_str());
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if (error == -1) {
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printf("ERROR in VideoEngine::SetTraceFile\n");
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return -1;
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}
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//
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// Create a VoE instance
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//
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webrtc::VoiceEngine* voe = webrtc::VoiceEngine::Create();
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//
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// Init VideoEngine and create a channel
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//
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webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
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if (ptrViEBase == NULL) {
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printf("ERROR in ViEBase::GetInterface\n");
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return -1;
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}
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error = ptrViEBase->Init();
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if (error == -1) {
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printf("ERROR in ViEBase::Init\n");
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return -1;
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}
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webrtc::VoEBase* voe_base = webrtc::VoEBase::GetInterface(voe);
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if (voe_base == NULL) {
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printf("ERROR in VoEBase::GetInterface\n");
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return -1;
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}
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error = voe_base->Init();
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if (error == -1) {
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printf("ERROR in VoEBase::Init\n");
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return -1;
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}
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int videoChannel = -1;
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error = ptrViEBase->CreateChannel(videoChannel);
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if (error == -1) {
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printf("ERROR in ViEBase::CreateChannel\n");
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return -1;
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}
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webrtc::VoEHardware* voe_hardware =
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webrtc::VoEHardware::GetInterface(voe);
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webrtc::VoECodec* voe_codec = webrtc::VoECodec::GetInterface(voe);
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webrtc::VoEAudioProcessing* voe_apm =
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webrtc::VoEAudioProcessing::GetInterface(voe);
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// Get the audio device for the call.
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memset(audio_capture_device_name, 0, KMaxUniqueIdLength);
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memset(audio_playbackDeviceName, 0, KMaxUniqueIdLength);
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GetAudioDevices(voe_base, voe_hardware, audio_capture_device_name,
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audio_capture_device_index, audio_playbackDeviceName,
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audio_playback_device_index);
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// Get the audio codec for the call.
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memset(static_cast<void*>(&audio_codec), 0, sizeof(audio_codec));
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GetAudioCodecRecord(voe_codec, audio_codec);
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audio_channel = voe_base->CreateChannel();
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error = voe_base->SetSendDestination(audio_channel, audio_tx_port,
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ipAddress);
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error = voe_base->SetLocalReceiver(audio_channel, audio_rx_port);
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error = voe_hardware->SetRecordingDevice(audio_capture_device_index);
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error = voe_hardware->SetPlayoutDevice(audio_playback_device_index);
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error = voe_codec->SetSendCodec(audio_channel, audio_codec);
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error = voe_apm->SetAgcStatus(true, webrtc::kAgcDefault);
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error = voe_apm->SetNsStatus(true, webrtc::kNsHighSuppression);
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//
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// List available capture devices, allocate and connect.
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//
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webrtc::ViECapture* ptrViECapture =
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webrtc::ViECapture::GetInterface(ptrViE);
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if (ptrViECapture == NULL) {
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printf("ERROR in ViECapture::GetInterface\n");
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return -1;
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}
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webrtc::VoERTP_RTCP* ptrVoERtpRtcp =
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webrtc::VoERTP_RTCP::GetInterface(voe);
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if (ptrVoERtpRtcp == NULL) {
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printf("ERROR in VoERTP_RTCP::GetInterface\n");
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return -1;
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}
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memset(deviceName, 0, KMaxDeviceNameLength);
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char uniqueId[KMaxUniqueIdLength];
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memset(uniqueId, 0, KMaxUniqueIdLength);
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printf("Available capture devices:\n");
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int captureIdx = 0;
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for (captureIdx = 0;
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captureIdx < ptrViECapture->NumberOfCaptureDevices();
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captureIdx++) {
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memset(deviceName, 0, KMaxDeviceNameLength);
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memset(uniqueId, 0, KMaxUniqueIdLength);
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error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
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KMaxDeviceNameLength, uniqueId,
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KMaxUniqueIdLength);
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if (error == -1) {
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printf("ERROR in ViECapture::GetCaptureDevice\n");
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return -1;
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}
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printf("\t %d. %s\n", captureIdx + 1, deviceName);
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}
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printf("\nChoose capture device: ");
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#ifdef WEBRTC_ANDROID
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captureIdx = 0;
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printf("0\n");
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#else
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if (scanf("%d", &captureIdx) != 1) {
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printf("Error in scanf()\n");
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return -1;
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}
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getchar();
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captureIdx = captureIdx - 1; // Compensate for idx start at 1.
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#endif
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error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
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KMaxDeviceNameLength, uniqueId,
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KMaxUniqueIdLength);
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if (error == -1) {
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printf("ERROR in ViECapture::GetCaptureDevice\n");
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return -1;
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}
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int captureId = 0;
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error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
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captureId);
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if (error == -1) {
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printf("ERROR in ViECapture::AllocateCaptureDevice\n");
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return -1;
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}
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error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
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if (error == -1) {
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printf("ERROR in ViECapture::ConnectCaptureDevice\n");
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return -1;
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}
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error = ptrViECapture->StartCapture(captureId);
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if (error == -1) {
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printf("ERROR in ViECapture::StartCapture\n");
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return -1;
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}
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//
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// RTP/RTCP settings
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//
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webrtc::ViERTP_RTCP* ptrViERtpRtcp =
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webrtc::ViERTP_RTCP::GetInterface(ptrViE);
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if (ptrViERtpRtcp == NULL) {
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printf("ERROR in ViERTP_RTCP::GetInterface\n");
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return -1;
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}
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error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
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webrtc::kRtcpCompound_RFC4585);
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if (error == -1) {
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printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
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return -1;
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}
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error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
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videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
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if (error == -1) {
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printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
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return -1;
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}
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error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
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if (error == -1) {
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printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
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return -1;
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}
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//
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// Set up rendering
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//
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webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
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if (ptrViERender == NULL) {
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printf("ERROR in ViERender::GetInterface\n");
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return -1;
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}
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error = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
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if (error == -1) {
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printf("ERROR in ViERender::AddRenderer\n");
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return -1;
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}
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error = ptrViERender->StartRender(captureId);
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if (error == -1) {
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printf("ERROR in ViERender::StartRender\n");
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return -1;
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}
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error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
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1.0);
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if (error == -1) {
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printf("ERROR in ViERender::AddRenderer\n");
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return -1;
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}
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error = ptrViERender->StartRender(videoChannel);
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if (error == -1) {
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printf("ERROR in ViERender::StartRender\n");
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return -1;
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}
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//
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// Setup codecs
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//
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webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
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if (ptrViECodec == NULL) {
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printf("ERROR in ViECodec::GetInterface\n");
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return -1;
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}
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webrtc::VideoCodec videoCodec;
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memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
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int codecIdx = 0;
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#ifdef WEBRTC_ANDROID
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codecIdx = 0;
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printf("0\n");
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#else
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codecIdx = 0; // Compensate for idx start at 1.
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#endif
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error = ptrViECodec->GetCodec(codecIdx, videoCodec);
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if (error == -1) {
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printf("ERROR in ViECodec::GetCodec\n");
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return -1;
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}
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// Set spatial resolution option
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videoCodec.width = DEFAULT_VIDEO_CODEC_WIDTH;
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videoCodec.height = DEFAULT_VIDEO_CODEC_HEIGHT;
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// Set start bit rate
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videoCodec.startBitrate = DEFAULT_VIDEO_CODEC_START_RATE;
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error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
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if (error == -1) {
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printf("ERROR in ViECodec::SetSendCodec\n");
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return -1;
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}
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//
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// Address settings
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//
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webrtc::ViENetwork* ptrViENetwork =
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webrtc::ViENetwork::GetInterface(ptrViE);
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if (ptrViENetwork == NULL) {
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printf("ERROR in ViENetwork::GetInterface\n");
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return -1;
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}
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// Setting External transport
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TbExternalTransport extTransport(*(ptrViENetwork));
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error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
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if (error == -1) {
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printf("ERROR in ViENetwork::SetLocalReceiver\n");
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return -1;
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}
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error = ptrViENetwork->SetSendDestination(videoChannel,
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ipAddress, rtpPort);
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if (error == -1) {
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printf("ERROR in ViENetwork::SetSendDestination\n");
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return -1;
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}
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std::string str;
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int enable_labeling = 0;
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std::cout << std::endl;
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std::cout << "Do you want to label this recording?" << std::endl;
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std::cout << "0. No (default)." << std::endl;
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std::cout << "1. This call will be labeled on the fly." << std::endl;
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std::getline(std::cin, str);
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enable_labeling = atoi(str.c_str());
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uint32_t folder_time = static_cast<uint32_t>
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(webrtc::TickTime::MillisecondTimestamp());
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std::stringstream folder_time_str;
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folder_time_str << folder_time;
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const std::string folder_name = "recording" + folder_time_str.str();
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printf("recording name = %s\n", folder_name.c_str());
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// TODO(mikhal): use file_utils.
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#ifdef WIN32
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_mkdir(folder_name.c_str());
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#else
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mkdir(folder_name.c_str(), 0777);
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#endif
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const std::string audio_filename = folder_name + DEFAULT_RECORDING_AUDIO;
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const std::string audio_rtp_filename = folder_name +
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DEFAULT_RECORDING_AUDIO_RTP;
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const std::string video_rtp_filename = folder_name +
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DEFAULT_RECORDING_VIDEO_RTP;
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std::fstream timing;
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if (enable_labeling == 1) {
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std::cout << "Press enter to stamp current time."<< std::endl;
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std::string timing_file = folder_name + "/labeling.txt";
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timing.open(timing_file.c_str(), std::fstream::out | std::fstream::app);
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}
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printf("\nPress enter to start recording\n");
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std::getline(std::cin, str);
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printf("\nRecording started\n\n");
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// Start record
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error = ptrViEBase->StartReceive(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViENetwork::StartReceive\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViEBase->StartSend(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViENetwork::StartSend\n");
|
||||
return -1;
|
||||
}
|
||||
error = voe_base->StartSend(audio_channel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in VoENetwork::StartSend\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
// Engine started
|
||||
|
||||
voe_apm->StartDebugRecording(audio_filename.c_str());
|
||||
ptrViERtpRtcp->StartRTPDump(videoChannel,
|
||||
video_rtp_filename.c_str(), webrtc::kRtpOutgoing);
|
||||
ptrVoERtpRtcp->StartRTPDump(audio_channel,
|
||||
audio_rtp_filename.c_str(), webrtc::kRtpOutgoing);
|
||||
printf("Press s + enter to stop...");
|
||||
int64_t clock_time;
|
||||
if (enable_labeling == 1) {
|
||||
clock_time = webrtc::TickTime::MillisecondTimestamp();
|
||||
timing << clock_time << std::endl;
|
||||
}
|
||||
char c = getchar();
|
||||
fflush(stdin);
|
||||
while (c != 's') {
|
||||
if (c == '\n' && enable_labeling == 1) {
|
||||
clock_time = webrtc::TickTime::MillisecondTimestamp();
|
||||
timing << clock_time << std::endl;
|
||||
}
|
||||
c = getchar();
|
||||
}
|
||||
if (enable_labeling == 1) {
|
||||
clock_time = webrtc::TickTime::MillisecondTimestamp();
|
||||
timing << clock_time << std::endl;
|
||||
}
|
||||
|
||||
ptrViERtpRtcp->StopRTPDump(videoChannel, webrtc::kRtpOutgoing);
|
||||
ptrVoERtpRtcp->StopRTPDump(audio_channel, webrtc::kRtpOutgoing);
|
||||
voe_apm->StopDebugRecording();
|
||||
if (enable_labeling == 1)
|
||||
timing.close();
|
||||
|
||||
// Testing finished. Tear down Video Engine
|
||||
|
||||
error = ptrViEBase->StopReceive(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViEBase::StopReceive\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViEBase->StopSend(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViEBase::StopSend\n");
|
||||
return -1;
|
||||
}
|
||||
error = voe_base->StopSend(audio_channel);
|
||||
|
||||
error = ptrViERender->StopRender(captureId);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViERender::StopRender\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViERender->RemoveRenderer(captureId);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViERender::RemoveRenderer\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViERender->StopRender(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViERender::StopRender\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViERender->RemoveRenderer(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViERender::RemoveRenderer\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViECapture->StopCapture(captureId);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViECapture::StopCapture\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViECapture->ReleaseCaptureDevice(captureId);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
error = ptrViEBase->DeleteChannel(videoChannel);
|
||||
if (error == -1) {
|
||||
printf("ERROR in ViEBase::DeleteChannel\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
int remainingInterfaces = 0;
|
||||
remainingInterfaces = ptrViECodec->Release();
|
||||
remainingInterfaces += ptrViECapture->Release();
|
||||
remainingInterfaces += ptrViERtpRtcp->Release();
|
||||
remainingInterfaces += ptrViERender->Release();
|
||||
remainingInterfaces += ptrViENetwork->Release();
|
||||
remainingInterfaces += ptrViEBase->Release();
|
||||
if (remainingInterfaces > 0) {
|
||||
printf("ERROR: Could not release all interfaces\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
bool deleted = webrtc::VideoEngine::Delete(ptrViE);
|
||||
if (deleted == false) {
|
||||
printf("ERROR in VideoEngine::Delete\n");
|
||||
return -1;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
// TODO(mikhal): Place above functionality under this class.
|
||||
int ViEAutoTest::ViERecordCall() {
|
||||
ViETest::Log(" ");
|
||||
ViETest::Log("========================================");
|
||||
ViETest::Log(" ViE Record Call\n");
|
||||
|
||||
if (VideoEngineSampleRecordCode(_window1, _window2) == 0) {
|
||||
ViETest::Log(" ");
|
||||
ViETest::Log(" ViE Autotest Record Call Done");
|
||||
ViETest::Log("========================================");
|
||||
ViETest::Log(" ");
|
||||
return 0;
|
||||
}
|
||||
|
||||
ViETest::Log(" ");
|
||||
ViETest::Log(" ViE Autotest Record Call Failed");
|
||||
ViETest::Log("========================================");
|
||||
ViETest::Log(" ");
|
||||
return 1;
|
||||
}
|
||||
|
||||
bool GetAudioCodecRecord(webrtc::VoECodec* voe_codec,
|
||||
webrtc::CodecInst& audio_codec) {
|
||||
int error = 0;
|
||||
int number_of_errors = 0;
|
||||
memset(&audio_codec, 0, sizeof(webrtc::CodecInst));
|
||||
|
||||
while (1) {
|
||||
int codec_idx = 0;
|
||||
int default_codec_idx = 0;
|
||||
for (codec_idx = 0; codec_idx < voe_codec->NumOfCodecs(); codec_idx++) {
|
||||
error = voe_codec->GetCodec(codec_idx, audio_codec);
|
||||
number_of_errors += ViETest::TestError(error == 0,
|
||||
"ERROR: %s at line %d",
|
||||
__FUNCTION__, __LINE__);
|
||||
|
||||
// Test for default codec index.
|
||||
if (strcmp(audio_codec.plname, DEFAULT_AUDIO_CODEC) == 0) {
|
||||
default_codec_idx = codec_idx;
|
||||
}
|
||||
}
|
||||
error = voe_codec->GetCodec(default_codec_idx, audio_codec);
|
||||
number_of_errors += ViETest::TestError(error == 0,
|
||||
"ERROR: %s at line %d",
|
||||
__FUNCTION__, __LINE__);
|
||||
return true;
|
||||
}
|
||||
assert(false);
|
||||
return false;
|
||||
}
|
||||
@ -77,6 +77,7 @@
|
||||
'source/vie_autotest_main.cc',
|
||||
'source/vie_autotest_network.cc',
|
||||
'source/vie_autotest_render.cc',
|
||||
'source/vie_autotest_record.cc',
|
||||
'source/vie_autotest_rtp_rtcp.cc',
|
||||
'source/vie_autotest_custom_call.cc',
|
||||
'source/vie_autotest_simulcast.cc',
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user