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-rw-r--r--chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc65
1 files changed, 65 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
new file mode 100644
index 00000000000..491196e0972
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
@@ -0,0 +1,65 @@
+/*
+ * 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 <sstream>
+
+#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 char* name,
+ int instance_index,
+ int reinit_index,
+ const std::string& suffix) {
+ std::stringstream ss;
+ ss << name << "_" << instance_index << "-" << reinit_index << suffix;
+ return ss.str();
+}
+#endif
+
+} // namespace
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+FILE* ApmDataDumper::GetRawFile(const char* name) {
+ std::string filename =
+ FormFileName(name, instance_index_, recording_set_index_, ".dat");
+ auto& f = raw_files_[filename];
+ if (!f) {
+ f.reset(fopen(filename.c_str(), "wb"));
+ }
+ return f.get();
+}
+
+WavWriter* ApmDataDumper::GetWavFile(const char* name,
+ int sample_rate_hz,
+ int num_channels) {
+ std::string filename =
+ FormFileName(name, instance_index_, recording_set_index_, ".wav");
+ auto& f = wav_files_[filename];
+ if (!f) {
+ f.reset(new WavWriter(filename.c_str(), sample_rate_hz, num_channels));
+ }
+ return f.get();
+}
+
+#endif
+
+} // namespace webrtc