AnotherFoxGuy/webrtc-audio-processing
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Make debugging bits optional

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Avoide the need to pull in protobuf and other related bits.
This commit is contained in:
Arun Raghavan 2011-09-15 13:11:39 +05:30
parent 2f65d90fa0
commit 4c87243593
1 changed files with 22 additions and 0 deletions
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22
src/modules/audio_processing/main/source/audio_processing_impl.cc
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@ -16,7 +16,9 @@
#include "critical_section_wrapper.h" #include "critical_section_wrapper.h"
#include "echo_cancellation_impl.h" #include "echo_cancellation_impl.h"
#include "echo_control_mobile_impl.h" #include "echo_control_mobile_impl.h"
#ifndef NDEBUG
#include "file_wrapper.h" #include "file_wrapper.h"
#endif
#include "high_pass_filter_impl.h" #include "high_pass_filter_impl.h"
#include "gain_control_impl.h" #include "gain_control_impl.h"
#include "level_estimator_impl.h" #include "level_estimator_impl.h"
@ -25,11 +27,13 @@
#include "processing_component.h" #include "processing_component.h"
#include "splitting_filter.h" #include "splitting_filter.h"
#include "voice_detection_impl.h" #include "voice_detection_impl.h"
#ifndef NDEBUG
#ifdef WEBRTC_ANDROID #ifdef WEBRTC_ANDROID
#include "external/webrtc/src/modules/audio_processing/main/source/debug.pb.h" #include "external/webrtc/src/modules/audio_processing/main/source/debug.pb.h"
#else #else
#include "webrtc/audio_processing/debug.pb.h" #include "webrtc/audio_processing/debug.pb.h"
#endif #endif
#endif /* NDEBUG */
namespace webrtc { namespace webrtc {
AudioProcessing* AudioProcessing::Create(int id) { AudioProcessing* AudioProcessing::Create(int id) {
@ -60,8 +64,10 @@ AudioProcessingImpl::AudioProcessingImpl(int id)
level_estimator_(NULL), level_estimator_(NULL),
noise_suppression_(NULL), noise_suppression_(NULL),
voice_detection_(NULL), voice_detection_(NULL),
#ifndef NDEBUG
debug_file_(FileWrapper::Create()), debug_file_(FileWrapper::Create()),
event_msg_(new audioproc::Event()), event_msg_(new audioproc::Event()),
#endif
crit_(CriticalSectionWrapper::CreateCriticalSection()), crit_(CriticalSectionWrapper::CreateCriticalSection()),
render_audio_(NULL), render_audio_(NULL),
capture_audio_(NULL), capture_audio_(NULL),
@ -104,6 +110,7 @@ AudioProcessingImpl::~AudioProcessingImpl() {
component_list_.pop_front(); component_list_.pop_front();
} }
#ifndef NDEBUG
if (debug_file_->Open()) { if (debug_file_->Open()) {
debug_file_->CloseFile(); debug_file_->CloseFile();
} }
@ -112,6 +119,7 @@ AudioProcessingImpl::~AudioProcessingImpl() {
delete event_msg_; delete event_msg_;
event_msg_ = NULL; event_msg_ = NULL;
#endif
delete crit_; delete crit_;
crit_ = NULL; crit_ = NULL;
@ -167,12 +175,14 @@ int AudioProcessingImpl::InitializeLocked() {
} }
} }
#ifndef NDEBUG
if (debug_file_->Open()) { if (debug_file_->Open()) {
int err = WriteInitMessage(); int err = WriteInitMessage();
if (err != kNoError) { if (err != kNoError) {
return err; return err;
} }
} }
#endif
return kNoError; return kNoError;
} }
@ -268,6 +278,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
return kBadDataLengthError; return kBadDataLengthError;
} }
#ifndef NDEBUG
if (debug_file_->Open()) { if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::STREAM); event_msg_->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = event_msg_->mutable_stream(); audioproc::Stream* msg = event_msg_->mutable_stream();
@ -279,6 +290,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
msg->set_drift(echo_cancellation_->stream_drift_samples()); msg->set_drift(echo_cancellation_->stream_drift_samples());
msg->set_level(gain_control_->stream_analog_level()); msg->set_level(gain_control_->stream_analog_level());
} }
#endif
capture_audio_->DeinterleaveFrom(frame); capture_audio_->DeinterleaveFrom(frame);
@ -358,6 +370,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
capture_audio_->InterleaveTo(frame); capture_audio_->InterleaveTo(frame);
#ifndef NDEBUG
if (debug_file_->Open()) { if (debug_file_->Open()) {
audioproc::Stream* msg = event_msg_->mutable_stream(); audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t data_size = sizeof(WebRtc_Word16) * const size_t data_size = sizeof(WebRtc_Word16) *
@ -369,6 +382,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
return err; return err;
} }
} }
#endif
return kNoError; return kNoError;
} }
@ -393,6 +407,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return kBadDataLengthError; return kBadDataLengthError;
} }
#ifndef NDEBUG
if (debug_file_->Open()) { if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::REVERSE_STREAM); event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream(); audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
@ -405,6 +420,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return err; return err;
} }
} }
#endif
render_audio_->DeinterleaveFrom(frame); render_audio_->DeinterleaveFrom(frame);
@ -471,6 +487,7 @@ bool AudioProcessingImpl::was_stream_delay_set() const {
int AudioProcessingImpl::StartDebugRecording( int AudioProcessingImpl::StartDebugRecording(
const char filename[AudioProcessing::kMaxFilenameSize]) { const char filename[AudioProcessing::kMaxFilenameSize]) {
#ifndef NDEBUG
CriticalSectionScoped crit_scoped(*crit_); CriticalSectionScoped crit_scoped(*crit_);
assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize); assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize);
@ -494,11 +511,13 @@ int AudioProcessingImpl::StartDebugRecording(
if (err != kNoError) { if (err != kNoError) {
return err; return err;
} }
#endif
return kNoError; return kNoError;
} }
int AudioProcessingImpl::StopDebugRecording() { int AudioProcessingImpl::StopDebugRecording() {
#ifndef NDEBUG
CriticalSectionScoped crit_scoped(*crit_); CriticalSectionScoped crit_scoped(*crit_);
// We just return if recording hasn't started. // We just return if recording hasn't started.
if (debug_file_->Open()) { if (debug_file_->Open()) {
@ -506,6 +525,7 @@ int AudioProcessingImpl::StopDebugRecording() {
return kFileError; return kFileError;
} }
} }
#endif
return kNoError; return kNoError;
} }
@ -605,6 +625,7 @@ WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
return kNoError; return kNoError;
} }
#ifndef NDEBUG
int AudioProcessingImpl::WriteMessageToDebugFile() { int AudioProcessingImpl::WriteMessageToDebugFile() {
int32_t size = event_msg_->ByteSize(); int32_t size = event_msg_->ByteSize();
if (size <= 0) { if (size <= 0) {
@ -648,4 +669,5 @@ int AudioProcessingImpl::WriteInitMessage() {
return kNoError; return kNoError;
} }
#endif
} // namespace webrtc } // namespace webrtc