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 |