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Adding the video debug api to vie test record

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Review URL: https://webrtc-codereview.appspot.com/763004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2681 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mikhal@webrtc.org 2012-08-28 20:55:10 +00:00
parent 12dbcd9798
commit 954cf806d9
1 changed files with 8 additions and 2 deletions
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10
src/video_engine/test/auto_test/source/vie_autotest_record.cc
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@ -25,6 +25,7 @@
#include "video_engine/include/vie_base.h"
#include "video_engine/include/vie_capture.h"
#include "video_engine/include/vie_codec.h"
#include "video_engine/include/vie_file.h"
#include "video_engine/include/vie_network.h"
#include "video_engine/include/vie_render.h"
#include "video_engine/include/vie_rtp_rtcp.h"
@ -40,6 +41,7 @@
#define DEFAULT_VIDEO_CODEC_START_RATE 1000
#define DEFAULT_RECORDING_FOLDER "RECORDING"
#define DEFAULT_RECORDING_AUDIO "/audio_debug.aec"
#define DEFAULT_RECORDING_VIDEO "/video_debug.yuv"
#define DEFAULT_RECORDING_AUDIO_RTP "/audio_rtpdump.rtp"
#define DEFAULT_RECORDING_VIDEO_RTP "/video_rtpdump.rtp"
@ -387,6 +389,7 @@ int VideoEngineSampleRecordCode(void* window1, void* window2) {
mkdir(folder_name.c_str(), 0777);
#endif
const std::string audio_filename = folder_name + DEFAULT_RECORDING_AUDIO;
const std::string video_filename = folder_name + DEFAULT_RECORDING_VIDEO;
const std::string audio_rtp_filename = folder_name +
DEFAULT_RECORDING_AUDIO_RTP;
const std::string video_rtp_filename = folder_name +
@ -401,7 +404,6 @@ int VideoEngineSampleRecordCode(void* window1, void* window2) {
std::getline(std::cin, str);
printf("\nRecording started\n\n");
// Start record
error = ptrViEBase->StartReceive(videoChannel);
if (error == -1) {
printf("ERROR in ViENetwork::StartReceive\n");
@ -419,9 +421,12 @@ int VideoEngineSampleRecordCode(void* window1, void* window2) {
return -1;
}
// Get file interface (video recording)
webrtc::ViEFile* vie_file = webrtc::ViEFile::GetInterface(ptrViE);
// Engine started
voe_apm->StartDebugRecording(audio_filename.c_str());
vie_file->StartDebugRecording(videoChannel, video_filename.c_str());
ptrViERtpRtcp->StartRTPDump(videoChannel,
video_rtp_filename.c_str(), webrtc::kRtpOutgoing);
ptrVoERtpRtcp->StartRTPDump(audio_channel,
@ -449,10 +454,11 @@ int VideoEngineSampleRecordCode(void* window1, void* window2) {
ptrViERtpRtcp->StopRTPDump(videoChannel, webrtc::kRtpOutgoing);
ptrVoERtpRtcp->StopRTPDump(audio_channel, webrtc::kRtpOutgoing);
voe_apm->StopDebugRecording();
vie_file->StopDebugRecording(videoChannel);
if (enable_labeling == 1)
timing.close();
// Testing finished. Tear down Video Engine
// Recording finished. Tear down Video Engine.
error = ptrViEBase->StopReceive(videoChannel);
if (error == -1) {