Close input file in FileAudioDevice::StopRecording.

webrtc/modules/audio_device/dummy/file_audio_device.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 
 namespace webrtc {
 
 const int kRecordingFixedSampleRate = 48000;
 const size_t kRecordingNumChannels = 2;
 const int kPlayoutFixedSampleRate = 48000;
 const size_t kPlayoutNumChannels = 2;
@@ skipping 171 matching lines @@
       _playoutBuffer = new int8_t[kPlayoutBufferSize];
   }
   if (!_playoutBuffer) {
     _playing = false;
     return -1;
   }
 
   // PLAYOUT
   if (!_outputFilename.empty() &&
       !_outputFile.OpenFile(_outputFilename.c_str(), false)) {
-    printf("Failed to open playout file %s!\n", _outputFilename.c_str());
+    LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
     _playing = false;
     delete [] _playoutBuffer;
     _playoutBuffer = NULL;
     return -1;
   }
 
   _ptrThreadPlay.reset(new rtc::PlatformThread(
       PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
   _ptrThreadPlay->Start();
   _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
+
+  LOG(LS_INFO) << "Started playout capture to output file: "
+               << _outputFilename;
   return 0;
 }
 
 int32_t FileAudioDevice::StopPlayout() {
   {
       CriticalSectionScoped lock(&_critSect);
       _playing = false;
   }
 
   // stop playout thread first
   if (_ptrThreadPlay) {
       _ptrThreadPlay->Stop();
       _ptrThreadPlay.reset();
   }
 
   CriticalSectionScoped lock(&_critSect);
 
   _playoutFramesLeft = 0;
   delete [] _playoutBuffer;
   _playoutBuffer = NULL;
   _outputFile.CloseFile();
+
+  LOG(LS_INFO) << "Stopped playout capture to output file: "
+               << _outputFilename;
   return 0;
 }
 
 bool FileAudioDevice::Playing() const {
   return true;
 }
 
 int32_t FileAudioDevice::StartRecording() {
   _recording = true;
 
   // Make sure we only create the buffer once.
   _recordingBufferSizeIn10MS = _recordingFramesIn10MS *
                                kRecordingNumChannels *
                                2;
   if (!_recordingBuffer) {
       _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
   }
 
   if (!_inputFilename.empty() &&
       !_inputFile.OpenFile(_inputFilename.c_str(), true)) {
-    printf("Failed to open audio input file %s!\n",
-           _inputFilename.c_str());
+    LOG(LS_ERROR) << "Failed to open audio input file: " << _inputFilename;
     _recording = false;
     delete[] _recordingBuffer;
     _recordingBuffer = NULL;
     return -1;
   }
 
   _ptrThreadRec.reset(new rtc::PlatformThread(
       RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
   _ptrThreadRec->Start();
   _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
+  LOG(LS_INFO) << "Started recording from input file: "
+               << _inputFilename;
+
   return 0;
 }
 
 
 int32_t FileAudioDevice::StopRecording() {
   {
     CriticalSectionScoped lock(&_critSect);
     _recording = false;
   }
 
   if (_ptrThreadRec) {
       _ptrThreadRec->Stop();
       _ptrThreadRec.reset();
   }
 
   CriticalSectionScoped lock(&_critSect);
   _recordingFramesLeft = 0;
   if (_recordingBuffer) {
       delete [] _recordingBuffer;
       _recordingBuffer = NULL;
   }
+  _inputFile.CloseFile();
+
+  LOG(LS_INFO) << "Stopped recording from input file: "
+               << _inputFilename;
   return 0;
 }
 
 bool FileAudioDevice::Recording() const {
   return _recording;
 }
 
 int32_t FileAudioDevice::SetAGC(bool enable) { return -1; }
 
 bool FileAudioDevice::AGC() const { return false; }
@@ skipping 228 matching lines @@
 
     uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
     if(deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc