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..491196e09729804af564af859146fdb08c9a1fe3 |
--- /dev/null |
+++ b/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 |