Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/
The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}
Review URL: https://codereview.webrtc.org/1540103002
Cr-Commit-Position: refs/heads/master@{#11267}
diff --git a/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
index 75ff245..eb4d959 100644
--- a/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
+++ b/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -498,7 +498,7 @@
ParcelFileDescriptor.MODE_READ_WRITE |
ParcelFileDescriptor.MODE_CREATE |
ParcelFileDescriptor.MODE_TRUNCATE);
- factory.startAecDump(aecDumpFileDescriptor.getFd());
+ factory.startAecDump(aecDumpFileDescriptor.getFd(), -1);
} catch(IOException e) {
Log.e(TAG, "Can not open aecdump file", e);
}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 744309c..0934b1e 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -647,6 +647,7 @@
++i)
msg->add_output_channel(dest[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+ &debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
}
#endif
@@ -734,6 +735,7 @@
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_output_data(frame->data_, data_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+ &debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
}
#endif
@@ -901,6 +903,7 @@
i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
msg->add_channel(src[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+ &debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.render));
}
#endif
@@ -969,6 +972,7 @@
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_data(frame->data_, data_size);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+ &debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.render));
}
#endif
@@ -1054,7 +1058,8 @@
}
int AudioProcessingImpl::StartDebugRecording(
- const char filename[AudioProcessing::kMaxFilenameSize]) {
+ const char filename[AudioProcessing::kMaxFilenameSize],
+ int64_t max_log_size_bytes) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
@@ -1065,6 +1070,7 @@
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
// Stop any ongoing recording.
if (debug_dump_.debug_file->Open()) {
if (debug_dump_.debug_file->CloseFile() == -1) {
@@ -1085,7 +1091,8 @@
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
-int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
+int AudioProcessingImpl::StartDebugRecording(FILE* handle,
+ int64_t max_log_size_bytes) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
@@ -1095,6 +1102,8 @@
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+ debug_dump_.num_bytes_left_for_log_ = max_log_size_bytes;
+
// Stop any ongoing recording.
if (debug_dump_.debug_file->Open()) {
if (debug_dump_.debug_file->CloseFile() == -1) {
@@ -1120,7 +1129,7 @@
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
- return StartDebugRecording(stream);
+ return StartDebugRecording(stream, -1);
}
int AudioProcessingImpl::StopDebugRecording() {
@@ -1416,6 +1425,7 @@
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
int AudioProcessingImpl::WriteMessageToDebugFile(
FileWrapper* debug_file,
+ int64_t* filesize_limit_bytes,
rtc::CriticalSection* crit_debug,
ApmDebugDumpThreadState* debug_state) {
int32_t size = debug_state->event_msg->ByteSize();
@@ -1433,7 +1443,19 @@
{
// Ensure atomic writes of the message.
- rtc::CritScope cs_capture(crit_debug);
+ rtc::CritScope cs_debug(crit_debug);
+
+ RTC_DCHECK(debug_file->Open());
+ // Update the byte counter.
+ if (*filesize_limit_bytes >= 0) {
+ *filesize_limit_bytes -=
+ (sizeof(int32_t) + debug_state->event_str.length());
+ if (*filesize_limit_bytes < 0) {
+ // Not enough bytes are left to write this message, so stop logging.
+ debug_file->CloseFile();
+ return kNoError;
+ }
+ }
// Write message preceded by its size.
if (!debug_file->Write(&size, sizeof(int32_t))) {
return kFileError;
@@ -1468,6 +1490,7 @@
// debug_dump_.capture.event_msg.
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+ &debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
return kNoError;
}
@@ -1520,6 +1543,7 @@
debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+ &debug_dump_.num_bytes_left_for_log_,
&crit_debug_, &debug_dump_.capture));
return kNoError;
}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index b310896..fbb9b6e 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -57,8 +57,10 @@
int Initialize(const ProcessingConfig& processing_config) override;
void SetExtraOptions(const Config& config) override;
void UpdateHistogramsOnCallEnd() override;
- int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
- int StartDebugRecording(FILE* handle) override;
+ int StartDebugRecording(const char filename[kMaxFilenameSize],
+ int64_t max_log_size_bytes) override;
+ int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override;
+
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
int StopDebugRecording() override;
@@ -144,6 +146,9 @@
struct ApmDebugDumpState {
ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
+ // Number of bytes that can still be written to the log before the maximum
+ // size is reached. A value of <= 0 indicates that no limit is used.
+ int64_t num_bytes_left_for_log_ = -1;
rtc::scoped_ptr<FileWrapper> debug_file;
ApmDebugDumpThreadState render;
ApmDebugDumpThreadState capture;
@@ -222,6 +227,7 @@
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
static int WriteMessageToDebugFile(FileWrapper* debug_file,
+ int64_t* filesize_limit_bytes,
rtc::CriticalSection* crit_debug,
ApmDebugDumpThreadState* debug_state);
int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 9a3a4b3..bf3565d 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -415,13 +415,22 @@
// Starts recording debugging information to a file specified by |filename|,
// a NULL-terminated string. If there is an ongoing recording, the old file
// will be closed, and recording will continue in the newly specified file.
- // An already existing file will be overwritten without warning.
+ // An already existing file will be overwritten without warning. A maximum
+ // file size (in bytes) for the log can be specified. The logging is stopped
+ // once the limit has been reached. If max_log_size_bytes is set to a value
+ // <= 0, no limit will be used.
static const size_t kMaxFilenameSize = 1024;
- virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
+ virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
+ int64_t max_log_size_bytes) = 0;
// Same as above but uses an existing file handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
- virtual int StartDebugRecording(FILE* handle) = 0;
+ virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
+
+ // TODO(ivoc): Remove this function after Chrome stops using it.
+ int StartDebugRecording(FILE* handle) {
+ return StartDebugRecording(handle, -1);
+ }
// Same as above but uses an existing PlatformFile handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
index 9e1f2d5..3a7e308 100644
--- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -250,10 +250,11 @@
void(int offset));
MOCK_CONST_METHOD0(delay_offset_ms,
int());
- MOCK_METHOD1(StartDebugRecording,
- int(const char filename[kMaxFilenameSize]));
- MOCK_METHOD1(StartDebugRecording,
- int(FILE* handle));
+ MOCK_METHOD2(StartDebugRecording,
+ int(const char filename[kMaxFilenameSize],
+ int64_t max_log_size_bytes));
+ MOCK_METHOD2(StartDebugRecording,
+ int(FILE* handle, int64_t max_log_size_bytes));
MOCK_METHOD0(StopDebugRecording,
int());
MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 94aea17..1b37120 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -383,7 +383,8 @@
int AnalyzeReverseStreamChooser(Format format);
void ProcessDebugDump(const std::string& in_filename,
const std::string& out_filename,
- Format format);
+ Format format,
+ int max_size_bytes);
void VerifyDebugDumpTest(Format format);
const std::string output_path_;
@@ -1706,7 +1707,8 @@
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
void ApmTest::ProcessDebugDump(const std::string& in_filename,
const std::string& out_filename,
- Format format) {
+ Format format,
+ int max_size_bytes) {
FILE* in_file = fopen(in_filename.c_str(), "rb");
ASSERT_TRUE(in_file != NULL);
audioproc::Event event_msg;
@@ -1734,7 +1736,8 @@
if (first_init) {
// StartDebugRecording() writes an additional init message. Don't start
// recording until after the first init to avoid the extra message.
- EXPECT_NOERR(apm_->StartDebugRecording(out_filename.c_str()));
+ EXPECT_NOERR(
+ apm_->StartDebugRecording(out_filename.c_str(), max_size_bytes));
first_init = false;
}
@@ -1809,34 +1812,54 @@
test::OutputPath(), std::string("ref") + format_string + "_aecdump");
const std::string out_filename = test::TempFilename(
test::OutputPath(), std::string("out") + format_string + "_aecdump");
+ const std::string limited_filename = test::TempFilename(
+ test::OutputPath(), std::string("limited") + format_string + "_aecdump");
+ const size_t logging_limit_bytes = 100000;
+ // We expect at least this many bytes in the created logfile.
+ const size_t logging_expected_bytes = 95000;
EnableAllComponents();
- ProcessDebugDump(in_filename, ref_filename, format);
- ProcessDebugDump(ref_filename, out_filename, format);
+ ProcessDebugDump(in_filename, ref_filename, format, -1);
+ ProcessDebugDump(ref_filename, out_filename, format, -1);
+ ProcessDebugDump(ref_filename, limited_filename, format, logging_limit_bytes);
FILE* ref_file = fopen(ref_filename.c_str(), "rb");
FILE* out_file = fopen(out_filename.c_str(), "rb");
+ FILE* limited_file = fopen(limited_filename.c_str(), "rb");
ASSERT_TRUE(ref_file != NULL);
ASSERT_TRUE(out_file != NULL);
+ ASSERT_TRUE(limited_file != NULL);
rtc::scoped_ptr<uint8_t[]> ref_bytes;
rtc::scoped_ptr<uint8_t[]> out_bytes;
+ rtc::scoped_ptr<uint8_t[]> limited_bytes;
size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
size_t out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
+ size_t limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
size_t bytes_read = 0;
+ size_t bytes_read_limited = 0;
while (ref_size > 0 && out_size > 0) {
bytes_read += ref_size;
+ bytes_read_limited += limited_size;
EXPECT_EQ(ref_size, out_size);
+ EXPECT_GE(ref_size, limited_size);
EXPECT_EQ(0, memcmp(ref_bytes.get(), out_bytes.get(), ref_size));
+ EXPECT_EQ(0, memcmp(ref_bytes.get(), limited_bytes.get(), limited_size));
ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
+ limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
}
EXPECT_GT(bytes_read, 0u);
+ EXPECT_GT(bytes_read_limited, logging_expected_bytes);
+ EXPECT_LE(bytes_read_limited, logging_limit_bytes);
EXPECT_NE(0, feof(ref_file));
EXPECT_NE(0, feof(out_file));
+ EXPECT_NE(0, feof(limited_file));
ASSERT_EQ(0, fclose(ref_file));
ASSERT_EQ(0, fclose(out_file));
+ ASSERT_EQ(0, fclose(limited_file));
remove(ref_filename.c_str());
remove(out_filename.c_str());
+ remove(limited_filename.c_str());
}
TEST_F(ApmTest, VerifyDebugDumpInt) {
@@ -1853,13 +1876,13 @@
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
EXPECT_EQ(apm_->kNullPointerError,
- apm_->StartDebugRecording(static_cast<const char*>(NULL)));
+ apm_->StartDebugRecording(static_cast<const char*>(NULL), -1));
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str()));
+ EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str(), -1));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
@@ -1873,7 +1896,7 @@
ASSERT_EQ(0, remove(filename.c_str()));
#else
EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(filename.c_str()));
+ apm_->StartDebugRecording(filename.c_str(), -1));
EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
// Verify the file has NOT been written.
@@ -1884,7 +1907,7 @@
// TODO(andrew): expand test to verify output.
TEST_F(ApmTest, DebugDumpFromFileHandle) {
FILE* fid = NULL;
- EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid));
+ EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid, -1));
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
fid = fopen(filename.c_str(), "w");
@@ -1894,7 +1917,7 @@
// Stopping without having started should be OK.
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid));
+ EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid, -1));
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
@@ -1908,7 +1931,7 @@
ASSERT_EQ(0, remove(filename.c_str()));
#else
EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(fid));
+ apm_->StartDebugRecording(fid, -1));
EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
ASSERT_EQ(0, fclose(fid));
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
index 005faa0..306e9e7 100644
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -181,7 +181,7 @@
}
void DebugDumpGenerator::StartRecording() {
- apm_->StartDebugRecording(dump_file_name_.c_str());
+ apm_->StartDebugRecording(dump_file_name_.c_str(), -1);
}
void DebugDumpGenerator::Process(size_t num_blocks) {
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 6e20a78..d987307 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -435,7 +435,7 @@
} else if (strcmp(argv[i], "--debug_file") == 0) {
i++;
ASSERT_LT(i, argc) << "Specify filename after --debug_file";
- ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i]));
+ ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i], -1));
} else {
FAIL() << "Unrecognized argument " << argv[i];
}
diff --git a/webrtc/voice_engine/voe_audio_processing_impl.cc b/webrtc/voice_engine/voe_audio_processing_impl.cc
index c957263..4479fe0 100644
--- a/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -924,7 +924,7 @@
return -1;
}
- return _shared->audio_processing()->StartDebugRecording(fileNameUTF8);
+ return _shared->audio_processing()->StartDebugRecording(fileNameUTF8, -1);
}
int VoEAudioProcessingImpl::StartDebugRecording(FILE* file_handle) {
@@ -935,7 +935,7 @@
return -1;
}
- return _shared->audio_processing()->StartDebugRecording(file_handle);
+ return _shared->audio_processing()->StartDebugRecording(file_handle, -1);
}
int VoEAudioProcessingImpl::StopDebugRecording() {
