From 4bc60452f73d430bc164cc73cc0c0028bf6cf153 Mon Sep 17 00:00:00 2001 From: Alessio Bazzica Date: Tue, 20 Nov 2018 12:44:15 +0100 Subject: [PATCH] Add output directory option for audioproc_f data dump files. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:10000 Change-Id: Iac21f826e78d6cb339c68fdeeedf9fe39920ac31 Reviewed-on: https://webrtc-review.googlesource.com/c/110904 Commit-Queue: Alessio Bazzica Reviewed-by: Per Åhgren Cr-Commit-Position: refs/heads/master@{#25713} --- modules/audio_processing/BUILD.gn | 1 + .../logging/apm_data_dumper.cc | 35 +++++++++++++------ .../logging/apm_data_dumper.h | 13 +++++++ .../test/audio_processing_simulator.cc | 4 +++ .../test/audio_processing_simulator.h | 1 + .../test/audioproc_float_impl.cc | 10 ++++++ 6 files changed, 54 insertions(+), 10 deletions(-) diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index dc1db11c76..4ea1e75ec9 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -329,6 +329,7 @@ rtc_source_set("apm_logging") { deps = [ "../../api:array_view", "../../common_audio:common_audio", + "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", "../../rtc_base:stringutils", ] diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc index 45d8088c98..cc879c8b08 100644 --- a/modules/audio_processing/logging/apm_data_dumper.cc +++ b/modules/audio_processing/logging/apm_data_dumper.cc @@ -19,16 +19,30 @@ #endif namespace webrtc { - namespace { #if WEBRTC_APM_DEBUG_DUMP == 1 -std::string FormFileName(const char* name, + +#if defined(WEBRTC_WIN) +constexpr char kPathDelimiter = '\\'; +#else +constexpr char kPathDelimiter = '/'; +#endif + +std::string FormFileName(const char* output_dir, + const char* name, int instance_index, int reinit_index, const std::string& suffix) { char buf[1024]; rtc::SimpleStringBuilder ss(buf); + const size_t output_dir_size = strlen(output_dir); + if (output_dir_size > 0) { + ss << output_dir; + if (output_dir[output_dir_size - 1] != kPathDelimiter) { + ss << kPathDelimiter; + } + } ss << name << "_" << instance_index << "-" << reinit_index << suffix; return ss.str(); } @@ -40,20 +54,22 @@ std::string FormFileName(const char* name, ApmDataDumper::ApmDataDumper(int instance_index) : instance_index_(instance_index) {} #else -ApmDataDumper::ApmDataDumper(int instance_index) {} +ApmDataDumper::ApmDataDumper(int instance_index){}; #endif -ApmDataDumper::~ApmDataDumper() {} +ApmDataDumper::~ApmDataDumper() = default; #if WEBRTC_APM_DEBUG_DUMP == 1 bool ApmDataDumper::recording_activated_ = false; -; +char ApmDataDumper::output_dir_[] = ""; + FILE* ApmDataDumper::GetRawFile(const char* name) { - std::string filename = - FormFileName(name, instance_index_, recording_set_index_, ".dat"); + std::string filename = FormFileName(output_dir_, name, instance_index_, + recording_set_index_, ".dat"); auto& f = raw_files_[filename]; if (!f) { f.reset(fopen(filename.c_str(), "wb")); + RTC_CHECK(f.get()) << "Cannot write to " << filename << "."; } return f.get(); } @@ -61,15 +77,14 @@ FILE* ApmDataDumper::GetRawFile(const char* name) { WavWriter* ApmDataDumper::GetWavFile(const char* name, int sample_rate_hz, int num_channels) { - std::string filename = - FormFileName(name, instance_index_, recording_set_index_, ".wav"); + std::string filename = FormFileName(output_dir_, 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 diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h index 5a8a3899cf..b541ae84b9 100644 --- a/modules/audio_processing/logging/apm_data_dumper.h +++ b/modules/audio_processing/logging/apm_data_dumper.h @@ -13,7 +13,9 @@ #include #include +#include +#include #if WEBRTC_APM_DEBUG_DUMP == 1 #include #endif @@ -21,6 +23,7 @@ #include "api/array_view.h" #if WEBRTC_APM_DEBUG_DUMP == 1 #include "common_audio/wav_file.h" +#include "rtc_base/checks.h" #endif #include "rtc_base/constructormagic.h" @@ -57,6 +60,14 @@ class ApmDataDumper { #endif } + // Set an optional output directory. + static void SetOutputDirectory(const std::string& output_dir) { +#if WEBRTC_APM_DEBUG_DUMP == 1 + RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength); + strncpy(output_dir_, output_dir.c_str(), output_dir.size()); +#endif + } + // Reinitializes the data dumping such that new versions // of all files being dumped to are created. void InitiateNewSetOfRecordings() { @@ -250,6 +261,8 @@ class ApmDataDumper { private: #if WEBRTC_APM_DEBUG_DUMP == 1 static bool recording_activated_; + static constexpr size_t kOutputDirMaxLength = 1024; + static char output_dir_[kOutputDirMaxLength]; const int instance_index_; int recording_set_index_ = 0; std::unordered_map> diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc index 601fcd71a1..df9f4976be 100644 --- a/modules/audio_processing/test/audio_processing_simulator.cc +++ b/modules/audio_processing/test/audio_processing_simulator.cc @@ -128,6 +128,10 @@ AudioProcessingSimulator::AudioProcessingSimulator( worker_queue_("file_writer_task_queue") { RTC_CHECK(!settings_.dump_internal_data || WEBRTC_APM_DEBUG_DUMP == 1); ApmDataDumper::SetActivated(settings_.dump_internal_data); + if (settings_.dump_internal_data_output_dir.has_value()) { + ApmDataDumper::SetOutputDirectory( + settings_.dump_internal_data_output_dir.value()); + } if (settings_.ed_graph_output_filename && !settings_.ed_graph_output_filename->empty()) { diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h index 2d109e59c6..4b67537b49 100644 --- a/modules/audio_processing/test/audio_processing_simulator.h +++ b/modules/audio_processing/test/audio_processing_simulator.h @@ -95,6 +95,7 @@ struct SimulationSettings { bool store_intermediate_output = false; bool print_aec3_parameter_values = false; bool dump_internal_data = false; + absl::optional dump_internal_data_output_dir; absl::optional custom_call_order_filename; absl::optional aec3_settings_filename; }; diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc index cce153e3f9..0ff2806e6d 100644 --- a/modules/audio_processing/test/audioproc_float_impl.cc +++ b/modules/audio_processing/test/audioproc_float_impl.cc @@ -216,6 +216,9 @@ WEBRTC_DEFINE_string(aec3_settings, WEBRTC_DEFINE_bool(dump_data, false, "Dump internal data during the call (requires build flag)"); +WEBRTC_DEFINE_string(dump_data_output_dir, + "", + "Internal data dump output directory"); WEBRTC_DEFINE_bool(help, false, "Print this message"); void SetSettingIfSpecified(const std::string& value, @@ -351,6 +354,8 @@ SimulationSettings CreateSettings() { settings.store_intermediate_output = FLAG_store_intermediate_output; settings.print_aec3_parameter_values = FLAG_print_aec3_parameter_values; settings.dump_internal_data = FLAG_dump_data; + SetSettingIfSpecified(FLAG_dump_data_output_dir, + &settings.dump_internal_data_output_dir); return settings; } @@ -505,6 +510,11 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) { ReportConditionalErrorAndExit( WEBRTC_APM_DEBUG_DUMP == 0 && settings.dump_internal_data, "Error: --dump_data cannot be set without proper build support.\n"); + + ReportConditionalErrorAndExit( + !settings.dump_internal_data && + settings.dump_internal_data_output_dir.has_value(), + "Error: --dump_data_output_dir cannot be set without --dump_data.\n"); } } // namespace