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Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.

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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}
This commit is contained in:
ivoc
2016-01-15 03:06:36 -08:00
committed by Commit bot
parent 74e8df81ae
commit d66b44d565
22 changed files with 142 additions and 60 deletions
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32
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -647,6 +647,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
++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 @@ 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
@ -901,6 +903,7 @@ 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
@ -969,6 +972,7 @@ 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
@ -1054,7 +1058,8 @@ 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_);
@ -1065,6 +1070,7 @@ 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) {
@ -1085,7 +1091,8 @@ 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_);
@ -1095,6 +1102,8 @@ 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) {
@ -1120,7 +1129,7 @@ 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() {
@ -1416,6 +1425,7 @@ 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();
@ -1433,7 +1443,19 @@ 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;
@ -1468,6 +1490,7 @@ 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;
}
@ -1520,6 +1543,7 @@ 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;
}