Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12a…

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 614 matching lines @@
   capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.output_stream().num_frames();
     for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++i)
       msg->add_output_channel(dest[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
   {
@@ skipping 67 matching lines @@
   capture_.capture_audio->InterleaveTo(frame,
                                        output_copy_needed(is_data_processed()));
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStreamLocked() {
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
@@ skipping 147 matching lines @@
   if (debug_dump_.debug_file->Open()) {
     debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
     audioproc::ReverseStream* msg =
         debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
     for (int i = 0;
          i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
       msg->add_channel(src[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
 
   render_.render_audio->CopyFrom(src,
                                  formats_.api_format.reverse_input_stream());
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
@@ skipping 48 matching lines @@
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
     audioproc::ReverseStream* msg =
         debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
   render_.render_audio->DeinterleaveFrom(frame);
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStreamLocked() {
   AudioBuffer* ra = render_.render_audio.get();  // For brevity.
   if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) {
@@ skipping 65 matching lines @@
   rtc::CritScope cs(&crit_capture_);
   capture_.delay_offset_ms = offset;
 }
 
 int AudioProcessingImpl::delay_offset_ms() const {
   rtc::CritScope cs(&crit_capture_);
   return capture_.delay_offset_ms;
 }
 
 int AudioProcessingImpl::StartDebugRecording(
-    const char filename[AudioProcessing::kMaxFilenameSize]) {
+    const char filename[AudioProcessing::kMaxFilenameSize],
+    int64_t max_log_size_bytes) {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
   static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
 
   if (filename == nullptr) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
   // Stop any ongoing recording.
   if (debug_dump_.debug_file->Open()) {
     if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
   if (debug_dump_.debug_file->OpenFile(filename, false) == -1) {
     debug_dump_.debug_file->CloseFile();
     return kFileError;
   }
 
   RETURN_ON_ERR(WriteConfigMessage(true));
   RETURN_ON_ERR(WriteInitMessage());
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
-int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
+int AudioProcessingImpl::StartDebugRecording(FILE* handle,
+                                             int64_t max_log_size_bytes) {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
 
   if (handle == nullptr) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
+
   // Stop any ongoing recording.
   if (debug_dump_.debug_file->Open()) {
     if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
   if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) {
     return kFileError;
   }
 
   RETURN_ON_ERR(WriteConfigMessage(true));
   RETURN_ON_ERR(WriteInitMessage());
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
-  return StartDebugRecording(stream);
+  return StartDebugRecording(stream, -1);
 }
 
 int AudioProcessingImpl::StopDebugRecording() {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
   if (debug_dump_.debug_file->Open()) {
@@ skipping 274 matching lines @@
     RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
                               capture_.aec_system_delay_jumps, 51);
   }
   capture_.aec_system_delay_jumps = -1;
   capture_.last_aec_system_delay_ms = 0;
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 int AudioProcessingImpl::WriteMessageToDebugFile(
     FileWrapper* debug_file,
+    int64_t* filesize_limit_bytes,
     rtc::CriticalSection* crit_debug,
     ApmDebugDumpThreadState* debug_state) {
   int32_t size = debug_state->event_msg->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
   }
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)
 // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
 //            pretty safe in assuming little-endian.
 #endif
 
   if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) {
     return kUnspecifiedError;
   }
 
   {
     // Ensure atomic writes of the message.
-    rtc::CritScope cs_capture(crit_debug);
+    rtc::CritScope cs_debug(crit_debug);
+
+    RTC_DCHECK(debug_file->Open());
+    // Update the byte counter.
+    if (*filesize_limit_bytes >= 0) {
+      *filesize_limit_bytes -=
+          (sizeof(int32_t) + debug_state->event_str.length());
+      if (*filesize_limit_bytes < 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;
     }
     if (!debug_file->Write(debug_state->event_str.data(),
                            debug_state->event_str.length())) {
       return kFileError;
     }
   }
 
@@ skipping 14 matching lines @@
   msg->set_num_reverse_channels(
       formats_.api_format.reverse_input_stream().num_channels());
   msg->set_reverse_sample_rate(
       formats_.api_format.reverse_input_stream().sample_rate_hz());
   msg->set_output_sample_rate(
       formats_.api_format.output_stream().sample_rate_hz());
   // TODO(ekmeyerson): Add reverse output fields to
   // debug_dump_.capture.event_msg.
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   audioproc::Config config;
 
   config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled());
   config.set_aec_delay_agnostic_enabled(
       public_submodules_->echo_cancellation->is_delay_agnostic_enabled());
@@ skipping 32 matching lines @@
       debug_dump_.capture.last_serialized_config == serialized_config) {
     return kNoError;
   }
 
   debug_dump_.capture.last_serialized_config = serialized_config;
 
   debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc