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Added option to specify a maximum file size when recording an AEC dump.

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For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.

BUG=webrtc:4741
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1413483003

Cr-Commit-Position: refs/heads/master@{#11081}
This commit is contained in:
ivoc
2015-12-18 03:53:37 -08:00
committed by Commit bot
parent 095ae15d6b
commit ae2c5ad12a
22 changed files with 136 additions and 60 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -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;
}