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Unified Diff: webrtc/modules/audio_processing/logging/apm_data_dumper.cc

Issue 1877713002: Replaced the data logging functionality in the AEC with a generic logging functionality (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Made the destructor out-of-line Created 4 years, 8 months ago
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Index: webrtc/modules/audio_processing/logging/apm_data_dumper.cc
diff --git a/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
new file mode 100644
index 0000000000000000000000000000000000000000..97233693a66c5f56e6b0cefbf4cee6e2536c3e81
--- /dev/null
+++ b/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2016 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/modules/audio_processing/logging/apm_data_dumper.h"
+
+#include "webrtc/base/stringutils.h"
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
+ (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
+#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+std::string FormFileName(const std::string& name,
+ int instance_index,
+ int reinit_index,
+ const std::string& suffix) {
+ char instance_index_string[10];
+ rtc::sprintfn(instance_index_string, sizeof(instance_index_string), "%d",
+ instance_index);
+ char reinit_index_string[10];
+ rtc::sprintfn(reinit_index_string, sizeof(reinit_index_string), "%d",
+ reinit_index);
+ return name + "_" + instance_index_string + "-" +
+ std::to_string(reinit_index) + suffix;
+}
+#endif
+
+} // namespace
+
+ApmDataDumper::~ApmDataDumper() {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+ for (auto& raw_files_element : raw_files_) {
+ fclose(raw_files_element.second);
+ }
+
+ // Deleting the wav files implicitly causes the files to be closed.
+ for (auto& wav_files_element : wav_files_) {
+ delete wav_files_element.second;
+ }
+#endif
+}
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+FILE* ApmDataDumper::GetRawFile(const std::string& name) {
+ std::string filename =
+ FormFileName(name, instance_index_, recording_set_index_, ".dat");
+ auto search = raw_files_.find(filename);
+ if (search != raw_files_.end()) {
+ return search->second;
+ }
+ FILE* file = fopen(filename.c_str(), "wb");
+ raw_files_[filename] = file;
+ return file;
+}
+
+WavWriter* ApmDataDumper::GetWavFile(const std::string& name,
+ int sample_rate_hz,
+ int num_channels) {
+ std::string filename =
+ FormFileName(name, instance_index_, recording_set_index_, ".wav");
+ auto search = wav_files_.find(filename);
+ if (search != wav_files_.end()) {
+ return search->second;
+ }
+ WavWriter* file =
+ new WavWriter(filename.c_str(), sample_rate_hz, num_channels);
+ wav_files_[filename] = file;
+ return file;
+}
+#endif
+
+} // namespace webrtc

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