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Unified Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1541633002: Reland "Added option to specify a maximum file size when recording an AEC dump." (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix to prevent Chrome from breaking. Created 5 years ago
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Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index a33294534307e1e593c1c605029eeb885d58f853..b79b4f0c7695d18e952ed7f06c7a324fc17ae90f 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -632,6 +632,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++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
@@ -719,6 +720,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
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
@@ -886,6 +888,7 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
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
@@ -954,6 +957,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
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
@@ -1039,7 +1043,8 @@ int AudioProcessingImpl::delay_offset_ms() const {
}
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_);
@@ -1050,6 +1055,7 @@ int AudioProcessingImpl::StartDebugRecording(
}
#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) {
@@ -1070,7 +1076,8 @@ int AudioProcessingImpl::StartDebugRecording(
#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_);
@@ -1080,6 +1087,8 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
}
#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) {
@@ -1105,7 +1114,7 @@ int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
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() {
@@ -1400,6 +1409,7 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
#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();
@@ -1417,7 +1427,19 @@ int AudioProcessingImpl::WriteMessageToDebugFile(
{
// 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;
@@ -1452,6 +1474,7 @@ int AudioProcessingImpl::WriteInitMessage() {
// 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;
}
@@ -1504,6 +1527,7 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) {
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;
}
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