Added option to specify a maximum file size when recording an AEC dump.

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index f3ee0a399d54eaefad5020a550816051c421d0eb..f658792f10ab0cd3f7336e9c063e303d70e63fb8 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -194,6 +194,7 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
       debug_file_(FileWrapper::Create()),
       event_msg_(new audioproc::Event()),
+      nr_bytes_left_for_log_(-1),
 #endif
       api_format_({{{kSampleRate16kHz, 1, false},
                     {kSampleRate16kHz, 1, false},
@@ -901,7 +902,8 @@ int AudioProcessingImpl::delay_offset_ms() const {
 }
 
 int AudioProcessingImpl::StartDebugRecording(
-    const char filename[AudioProcessing::kMaxFilenameSize]) {
+    const char filename[AudioProcessing::kMaxFilenameSize],
+    int64_t max_log_size_bytes) {
   CriticalSectionScoped crit_scoped(crit_);
   static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
 
@@ -910,13 +912,8 @@ int AudioProcessingImpl::StartDebugRecording(
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  nr_bytes_left_for_log_ = max_log_size_bytes;
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {

[Andrew MacDonald, 2015/11/24 17:28:59]

Why did you remove this block? It's not removed from the overloaded method
below.

[ivoc, 2015/12/01 15:17:16]

Good point, I'm not sure why I removed it initially, so I will put it back.

-    if (debug_file_->CloseFile() == -1) {
-      return kFileError;
-    }
-  }
-
   if (debug_file_->OpenFile(filename, false) == -1) {
     debug_file_->CloseFile();
     return kFileError;
@@ -930,7 +927,8 @@ int AudioProcessingImpl::StartDebugRecording(
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
-int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
+int AudioProcessingImpl::StartDebugRecording(FILE* handle,
+                                             int64_t max_log_size_bytes) {
   CriticalSectionScoped crit_scoped(crit_);
 
   if (handle == NULL) {
@@ -938,6 +936,8 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  nr_bytes_left_for_log_ = max_log_size_bytes;
+
   // Stop any ongoing recording.
   if (debug_file_->Open()) {
     if (debug_file_->CloseFile() == -1) {
@@ -960,7 +960,7 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
-  return StartDebugRecording(stream);
+  return StartDebugRecording(stream, -1);
 }
 
 int AudioProcessingImpl::StopDebugRecording() {
@@ -1181,6 +1181,7 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 int AudioProcessingImpl::WriteMessageToDebugFile() {
+  RTC_DCHECK(debug_file_->Open());
   int32_t size = event_msg_->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
@@ -1194,6 +1195,15 @@ int AudioProcessingImpl::WriteMessageToDebugFile() {
     return kUnspecifiedError;
   }
 
+  if (nr_bytes_left_for_log_ >= 0) {
+    nr_bytes_left_for_log_ -= (sizeof(int32_t) + event_str_.length());
+    if (nr_bytes_left_for_log_ < 0) {
+      // Not enough bytes are left to write this message, so stop logging.
+      debug_file_->CloseFile();
+      return kNoError;
+    }
+  }
+
   // Write message preceded by its size.
   if (!debug_file_->Write(&size, sizeof(int32_t))) {
     return kFileError;