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Update the dummy file_audio_video_device to allow empty file name

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Landing this on behalf of malmnas@.

The semantics is as follows:

* if the output filename is empty, then don't log to file
* if the input filename is empty, then don't stream any audio from file

This is useful for long running tests with multiple participants.
With logging turned on, having 10 bots running for 2 hours results in
more then 7 GB of data.

BUG=None
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41219004

Cr-Commit-Position: refs/heads/master@{#8691}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8691 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org 2015-03-12 10:27:52 +00:00
parent e9413c686e
commit 5c9f69f0af
3 changed files with 44 additions and 46 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

83
webrtc/modules/audio_device/dummy/file_audio_device.cc
View File

@ -25,7 +25,7 @@ int kRecordingBufferSize = kRecordingFixedSampleRate / 100
FileAudioDevice::FileAudioDevice(const int32_t id,
const char* inputFilename,
const char* outputFile):
const char* outputFilename):
_ptrAudioBuffer(NULL),
_recordingBuffer(NULL),
_playoutBuffer(NULL),
@ -45,17 +45,21 @@ FileAudioDevice::FileAudioDevice(const int32_t id,
_lastCallRecordMillis(0),
_outputFile(*FileWrapper::Create()),
_inputFile(*FileWrapper::Create()),
_outputFilename(outputFile),
_outputFilename(outputFilename),
_inputFilename(inputFilename),
_clock(Clock::GetRealTimeClock()) {
}
FileAudioDevice::~FileAudioDevice() {
_outputFile.Flush();
_outputFile.CloseFile();
if (_outputFile.Open()) {
_outputFile.Flush();
_outputFile.CloseFile();
}
delete &_outputFile;
_inputFile.Flush();
_inputFile.CloseFile();
if (_inputFile.Open()) {
_inputFile.Flush();
_inputFile.CloseFile();
}
delete &_inputFile;
}
@ -144,8 +148,7 @@ int32_t FileAudioDevice::PlayoutIsAvailable(bool& available) {
}
int32_t FileAudioDevice::InitPlayout() {
if (_ptrAudioBuffer)
{
if (_ptrAudioBuffer) {
// Update webrtc audio buffer with the selected parameters
_ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
_ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
@ -187,21 +190,20 @@ bool FileAudioDevice::RecordingIsInitialized() const {
}
int32_t FileAudioDevice::StartPlayout() {
if (_playing)
{
if (_playing) {
return 0;
}
_playoutFramesIn10MS = kPlayoutFixedSampleRate/100;
_playing = true;
_playoutFramesLeft = 0;
_playoutFramesIn10MS = kPlayoutFixedSampleRate/100;
if (!_playoutBuffer)
if (!_playoutBuffer) {
_playoutBuffer = new int8_t[2 *
kPlayoutNumChannels *
kPlayoutFixedSampleRate/100];
if (!_playoutBuffer)
{
}
if (!_playoutBuffer) {
_playing = false;
return -1;
}
@ -212,17 +214,16 @@ int32_t FileAudioDevice::StartPlayout() {
this,
kRealtimePriority,
threadName);
if (_ptrThreadPlay == NULL)
{
if (_ptrThreadPlay == NULL) {
_playing = false;
delete [] _playoutBuffer;
_playoutBuffer = NULL;
return -1;
}
if (_outputFile.OpenFile(_outputFilename.c_str(),
false, false, false) == -1) {
printf("Failed to open playout file %s!", _outputFilename.c_str());
if (!_outputFilename.empty() && _outputFile.OpenFile(
_outputFilename.c_str(), false, false, false) == -1) {
printf("Failed to open playout file %s!\n", _outputFilename.c_str());
_playing = false;
delete [] _playoutBuffer;
_playoutBuffer = NULL;
@ -230,8 +231,7 @@ int32_t FileAudioDevice::StartPlayout() {
}
unsigned int threadID(0);
if (!_ptrThreadPlay->Start(threadID))
{
if (!_ptrThreadPlay->Start(threadID)) {
_playing = false;
delete _ptrThreadPlay;
_ptrThreadPlay = NULL;
@ -251,11 +251,9 @@ int32_t FileAudioDevice::StopPlayout() {
}
// stop playout thread first
if (_ptrThreadPlay && !_ptrThreadPlay->Stop())
{
if (_ptrThreadPlay && !_ptrThreadPlay->Stop()) {
return -1;
}
else {
} else {
delete _ptrThreadPlay;
_ptrThreadPlay = NULL;
}
@ -265,8 +263,10 @@ int32_t FileAudioDevice::StopPlayout() {
_playoutFramesLeft = 0;
delete [] _playoutBuffer;
_playoutBuffer = NULL;
_outputFile.Flush();
_outputFile.CloseFile();
if (_outputFile.Open()) {
_outputFile.Flush();
_outputFile.CloseFile();
}
return 0;
}
@ -285,8 +285,8 @@ int32_t FileAudioDevice::StartRecording() {
_recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
}
if (_inputFile.OpenFile(_inputFilename.c_str(), true,
true, false) == -1) {
if (!_inputFilename.empty() && _inputFile.OpenFile(
_inputFilename.c_str(), true, true, false) == -1) {
printf("Failed to open audio input file %s!\n",
_inputFilename.c_str());
_recording = false;
@ -300,8 +300,7 @@ int32_t FileAudioDevice::StartRecording() {
this,
kRealtimePriority,
threadName);
if (_ptrThreadRec == NULL)
{
if (_ptrThreadRec == NULL) {
_recording = false;
delete [] _recordingBuffer;
_recordingBuffer = NULL;
@ -309,8 +308,7 @@ int32_t FileAudioDevice::StartRecording() {
}
unsigned int threadID(0);
if (!_ptrThreadRec->Start(threadID))
{
if (!_ptrThreadRec->Start(threadID)) {
_recording = false;
delete _ptrThreadRec;
_ptrThreadRec = NULL;
@ -330,19 +328,16 @@ int32_t FileAudioDevice::StopRecording() {
_recording = false;
}
if (_ptrThreadRec && !_ptrThreadRec->Stop())
{
if (_ptrThreadRec && !_ptrThreadRec->Stop()) {
return -1;
}
else {
} else {
delete _ptrThreadRec;
_ptrThreadRec = NULL;
}
CriticalSectionScoped lock(&_critSect);
_recordingFramesLeft = 0;
if (_recordingBuffer)
{
if (_recordingBuffer) {
delete [] _recordingBuffer;
_recordingBuffer = NULL;
}
@ -528,15 +523,14 @@ bool FileAudioDevice::RecThreadFunc(void* pThis)
bool FileAudioDevice::PlayThreadProcess()
{
if(!_playing)
if(!_playing) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
_critSect.Enter();
if (_lastCallPlayoutMillis == 0 ||
currentTime - _lastCallPlayoutMillis >= 10)
{
currentTime - _lastCallPlayoutMillis >= 10) {
_critSect.Leave();
_ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
_critSect.Enter();
@ -557,8 +551,9 @@ bool FileAudioDevice::PlayThreadProcess()
bool FileAudioDevice::RecThreadProcess()
{
if (!_recording)
if (!_recording) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
_critSect.Enter();

6
webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
View File

@ -16,13 +16,14 @@
namespace webrtc {
bool FileAudioDeviceFactory::_isConfigured = false;
char FileAudioDeviceFactory::_inputAudioFilename[MAX_FILENAME_LEN] = "";
char FileAudioDeviceFactory::_outputAudioFilename[MAX_FILENAME_LEN] = "";
FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice(
const int32_t id) {
// Bail out here if the files aren't set.
if (strlen(_inputAudioFilename) == 0 || strlen(_outputAudioFilename) == 0) {
// Bail out here if the files haven't been set explicitly.
if (!_isConfigured) {
printf("Was compiled with WEBRTC_DUMMY_AUDIO_PLAY_STATIC_FILE "
"but did not set input/output files to use. Bailing out.\n");
exit(1);
@ -39,6 +40,7 @@ void FileAudioDeviceFactory::SetFilenamesToUse(
// Copy the strings since we don't know the lifetime of the input pointers.
strncpy(_inputAudioFilename, inputAudioFilename, MAX_FILENAME_LEN);
strncpy(_outputAudioFilename, outputAudioFilename, MAX_FILENAME_LEN);
_isConfigured = true;
#else
// Sanity: must be compiled with the right define to run this.
printf("Trying to use dummy file devices, but is not compiled "

1
webrtc/modules/audio_device/dummy/file_audio_device_factory.h
View File

@ -32,6 +32,7 @@ class FileAudioDeviceFactory {
private:
static const uint32_t MAX_FILENAME_LEN = 256;
static bool _isConfigured;
static char _inputAudioFilename[MAX_FILENAME_LEN];
static char _outputAudioFilename[MAX_FILENAME_LEN];
};