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Arun Raghavan
webrtc-audio-processing
Commits
4c872435
Commit
4c872435
authored
Sep 15, 2011
by
Arun Raghavan
Browse files
Make debugging bits optional
Avoide the need to pull in protobuf and other related bits.
parent
2f65d90f
Changes
1
Hide whitespace changes
Inline
Side-by-side
src/modules/audio_processing/main/source/audio_processing_impl.cc
View file @
4c872435
...
...
@@ -16,7 +16,9 @@
#include
"critical_section_wrapper.h"
#include
"echo_cancellation_impl.h"
#include
"echo_control_mobile_impl.h"
#ifndef NDEBUG
#include
"file_wrapper.h"
#endif
#include
"high_pass_filter_impl.h"
#include
"gain_control_impl.h"
#include
"level_estimator_impl.h"
...
...
@@ -25,11 +27,13 @@
#include
"processing_component.h"
#include
"splitting_filter.h"
#include
"voice_detection_impl.h"
#ifndef NDEBUG
#ifdef WEBRTC_ANDROID
#include
"external/webrtc/src/modules/audio_processing/main/source/debug.pb.h"
#else
#include
"webrtc/audio_processing/debug.pb.h"
#endif
#endif
/* NDEBUG */
namespace
webrtc
{
AudioProcessing
*
AudioProcessing
::
Create
(
int
id
)
{
...
...
@@ -60,8 +64,10 @@ AudioProcessingImpl::AudioProcessingImpl(int id)
level_estimator_
(
NULL
),
noise_suppression_
(
NULL
),
voice_detection_
(
NULL
),
#ifndef NDEBUG
debug_file_
(
FileWrapper
::
Create
()),
event_msg_
(
new
audioproc
::
Event
()),
#endif
crit_
(
CriticalSectionWrapper
::
CreateCriticalSection
()),
render_audio_
(
NULL
),
capture_audio_
(
NULL
),
...
...
@@ -104,6 +110,7 @@ AudioProcessingImpl::~AudioProcessingImpl() {
component_list_
.
pop_front
();
}
#ifndef NDEBUG
if
(
debug_file_
->
Open
())
{
debug_file_
->
CloseFile
();
}
...
...
@@ -112,6 +119,7 @@ AudioProcessingImpl::~AudioProcessingImpl() {
delete
event_msg_
;
event_msg_
=
NULL
;
#endif
delete
crit_
;
crit_
=
NULL
;
...
...
@@ -167,12 +175,14 @@ int AudioProcessingImpl::InitializeLocked() {
}
}
#ifndef NDEBUG
if
(
debug_file_
->
Open
())
{
int
err
=
WriteInitMessage
();
if
(
err
!=
kNoError
)
{
return
err
;
}
}
#endif
return
kNoError
;
}
...
...
@@ -268,6 +278,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
return
kBadDataLengthError
;
}
#ifndef NDEBUG
if
(
debug_file_
->
Open
())
{
event_msg_
->
set_type
(
audioproc
::
Event
::
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_level
(
gain_control_
->
stream_analog_level
());
}
#endif
capture_audio_
->
DeinterleaveFrom
(
frame
);
...
...
@@ -358,6 +370,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
capture_audio_
->
InterleaveTo
(
frame
);
#ifndef NDEBUG
if
(
debug_file_
->
Open
())
{
audioproc
::
Stream
*
msg
=
event_msg_
->
mutable_stream
();
const
size_t
data_size
=
sizeof
(
WebRtc_Word16
)
*
...
...
@@ -369,6 +382,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
return
err
;
}
}
#endif
return
kNoError
;
}
...
...
@@ -393,6 +407,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return
kBadDataLengthError
;
}
#ifndef NDEBUG
if
(
debug_file_
->
Open
())
{
event_msg_
->
set_type
(
audioproc
::
Event
::
REVERSE_STREAM
);
audioproc
::
ReverseStream
*
msg
=
event_msg_
->
mutable_reverse_stream
();
...
...
@@ -405,6 +420,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return
err
;
}
}
#endif
render_audio_
->
DeinterleaveFrom
(
frame
);
...
...
@@ -471,6 +487,7 @@ bool AudioProcessingImpl::was_stream_delay_set() const {
int
AudioProcessingImpl
::
StartDebugRecording
(
const
char
filename
[
AudioProcessing
::
kMaxFilenameSize
])
{
#ifndef NDEBUG
CriticalSectionScoped
crit_scoped
(
*
crit_
);
assert
(
kMaxFilenameSize
==
FileWrapper
::
kMaxFileNameSize
);
...
...
@@ -494,11 +511,13 @@ int AudioProcessingImpl::StartDebugRecording(
if
(
err
!=
kNoError
)
{
return
err
;
}
#endif
return
kNoError
;
}
int
AudioProcessingImpl
::
StopDebugRecording
()
{
#ifndef NDEBUG
CriticalSectionScoped
crit_scoped
(
*
crit_
);
// We just return if recording hasn't started.
if
(
debug_file_
->
Open
())
{
...
...
@@ -506,6 +525,7 @@ int AudioProcessingImpl::StopDebugRecording() {
return
kFileError
;
}
}
#endif
return
kNoError
;
}
...
...
@@ -605,6 +625,7 @@ WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
return
kNoError
;
}
#ifndef NDEBUG
int
AudioProcessingImpl
::
WriteMessageToDebugFile
()
{
int32_t
size
=
event_msg_
->
ByteSize
();
if
(
size
<=
0
)
{
...
...
@@ -648,4 +669,5 @@ int AudioProcessingImpl::WriteInitMessage() {
return
kNoError
;
}
#endif
}
// namespace webrtc
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