Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> 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.
>
> NOTRY=true
> 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}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index 52b6749..c31859d 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -1280,12 +1280,11 @@
}
JOW(jboolean, PeerConnectionFactory_nativeStartAecDump)(
- JNIEnv* jni, jclass, jlong native_factory, jint file,
- jint filesize_limit_bytes) {
+ JNIEnv* jni, jclass, jlong native_factory, jint file) {
#if defined(ANDROID)
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
factoryFromJava(native_factory));
- return factory->StartAecDump(file, filesize_limit_bytes);
+ return factory->StartAecDump(file);
#else
return false;
#endif
diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java b/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
index e65c736..d759c69 100644
--- a/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
+++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
@@ -143,8 +143,8 @@
// Starts recording an AEC dump. Ownership of the file is transfered to the
// native code. If an AEC dump is already in progress, it will be stopped and
// a new one will start using the provided file.
- public boolean startAecDump(int file_descriptor, int filesize_limit_bytes) {
- return nativeStartAecDump(nativeFactory, file_descriptor, filesize_limit_bytes);
+ public boolean startAecDump(int file_descriptor) {
+ return nativeStartAecDump(nativeFactory, file_descriptor);
}
// Stops recording an AEC dump. If no AEC dump is currently being recorded,
@@ -250,8 +250,7 @@
private static native long nativeCreateAudioTrack(
long nativeFactory, String id, long nativeSource);
- private static native boolean nativeStartAecDump(
- long nativeFactory, int file_descriptor, int filesize_limit_bytes);
+ private static native boolean nativeStartAecDump(long nativeFactory, int file_descriptor);
private static native void nativeStopAecDump(long nativeFactory);
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index 872bdb6..6d36c8b 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -232,10 +232,9 @@
return VideoSourceProxy::Create(signaling_thread_, source);
}
-bool PeerConnectionFactory::StartAecDump(rtc::PlatformFile file,
- int64_t max_size_bytes) {
+bool PeerConnectionFactory::StartAecDump(rtc::PlatformFile file) {
RTC_DCHECK(signaling_thread_->IsCurrent());
- return channel_manager_->StartAecDump(file, max_size_bytes);
+ return channel_manager_->StartAecDump(file);
}
void PeerConnectionFactory::StopAecDump() {
diff --git a/talk/app/webrtc/peerconnectionfactory.h b/talk/app/webrtc/peerconnectionfactory.h
index 6d2d972..cad89d4 100644
--- a/talk/app/webrtc/peerconnectionfactory.h
+++ b/talk/app/webrtc/peerconnectionfactory.h
@@ -93,7 +93,7 @@
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source) override;
- bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
+ bool StartAecDump(rtc::PlatformFile file) override;
void StopAecDump() override;
bool StartRtcEventLog(rtc::PlatformFile file) override;
void StopRtcEventLog() override;
diff --git a/talk/app/webrtc/peerconnectionfactoryproxy.h b/talk/app/webrtc/peerconnectionfactoryproxy.h
index f54ce5b..db34ea7 100644
--- a/talk/app/webrtc/peerconnectionfactoryproxy.h
+++ b/talk/app/webrtc/peerconnectionfactoryproxy.h
@@ -72,7 +72,7 @@
CreateVideoTrack, const std::string&, VideoSourceInterface*)
PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
CreateAudioTrack, const std::string&, AudioSourceInterface*)
- PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
+ PROXY_METHOD1(bool, StartAecDump, rtc::PlatformFile)
PROXY_METHOD0(void, StopAecDump)
PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
PROXY_METHOD0(void, StopRtcEventLog)
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 475145a..4648176 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -640,11 +640,9 @@
// Starts AEC dump using existing file. Takes ownership of |file| and passes
// it on to VoiceEngine (via other objects) immediately, which will take
// the ownerhip. If the operation fails, the file will be closed.
- // A maximum file size in bytes can be specified. When the file size limit is
- // reached, logging is stopped automatically. If max_size_bytes is set to a
- // value <= 0, no limit will be used, and logging will continue until the
- // StopAecDump function is called.
- virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
+ // TODO(grunell): Remove when Chromium has started to use AEC in each source.
+ // http://crbug.com/264611.
+ virtual bool StartAecDump(rtc::PlatformFile file) = 0;
// Stops logging the AEC dump.
virtual void StopAecDump() = 0;
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index 3f56512..f5b2174 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -762,9 +762,7 @@
int GetInputLevel() { return 0; }
- bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
- return false;
- }
+ bool StartAecDump(rtc::PlatformFile file) { return false; }
void StopAecDump() {}
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index cd533c4..43b4de5 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -102,10 +102,8 @@
virtual const std::vector<VideoCodec>& video_codecs() = 0;
virtual RtpCapabilities GetVideoCapabilities() = 0;
- // Starts AEC dump using existing file, a maximum file size in bytes can be
- // specified. Logging is stopped just before the size limit is exceeded.
- // If max_size_bytes is set to a value <= 0, no limit will be used.
- virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
+ // Starts AEC dump using existing file.
+ virtual bool StartAecDump(rtc::PlatformFile file) = 0;
// Stops recording AEC dump.
virtual void StopAecDump() = 0;
@@ -187,8 +185,8 @@
return video_.GetCapabilities();
}
- virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
- return voice_.StartAecDump(file, max_size_bytes);
+ virtual bool StartAecDump(rtc::PlatformFile file) {
+ return voice_.StartAecDump(file);
}
virtual void StopAecDump() {
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index debaa6e..bf22a29 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -113,9 +113,8 @@
WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
WEBRTC_STUB_CONST(delay_offset_ms, ());
- WEBRTC_STUB(StartDebugRecording,
- (const char filename[kMaxFilenameSize], int64_t max_size_bytes));
- WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes));
+ WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
+ WEBRTC_STUB(StartDebugRecording, (FILE* handle));
WEBRTC_STUB(StopDebugRecording, ());
WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 38b6c54..9eee2af 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1011,8 +1011,7 @@
return true;
}
-bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
- int64_t max_size_bytes) {
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
if (!aec_dump_file_stream) {
@@ -1022,8 +1021,7 @@
return false;
}
StopAecDump();
- if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
- aec_dump_file_stream, max_size_bytes) !=
+ if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StartDebugRecording);
fclose(aec_dump_file_stream);
@@ -1037,8 +1035,8 @@
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!is_dumping_aec_) {
// Start dumping AEC when we are not dumping.
- if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
- filename.c_str(), -1) != webrtc::AudioProcessing::kNoError) {
+ if (voe_wrapper_->processing()->StartDebugRecording(
+ filename.c_str()) != webrtc::AudioProcessing::kNoError) {
LOG_RTCERR1(StartDebugRecording, filename.c_str());
} else {
is_dumping_aec_ = true;
@@ -1050,7 +1048,7 @@
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (is_dumping_aec_) {
// Stop dumping AEC when we are dumping.
- if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
+ if (voe_wrapper_->processing()->StopDebugRecording() !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StopDebugRecording);
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index ce3bdf3..0f2f59e 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -94,11 +94,8 @@
// Set the external ADM. This can only be called before Init.
bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
- // Starts AEC dump using an existing file. A maximum file size in bytes can be
- // specified. When the maximum file size is reached, logging is stopped and
- // the file is closed. If max_size_bytes is set to <= 0, no limit will be
- // used.
- bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
+ // Starts AEC dump using existing file.
+ bool StartAecDump(rtc::PlatformFile file);
// Stops AEC dump.
void StopAecDump();
diff --git a/talk/session/media/channelmanager.cc b/talk/session/media/channelmanager.cc
index bd89a41..e7a4b8b 100644
--- a/talk/session/media/channelmanager.cc
+++ b/talk/session/media/channelmanager.cc
@@ -550,11 +550,9 @@
}
}
-bool ChannelManager::StartAecDump(rtc::PlatformFile file,
- int64_t max_size_bytes) {
- return worker_thread_->Invoke<bool>(Bind(&MediaEngineInterface::StartAecDump,
- media_engine_.get(), file,
- max_size_bytes));
+bool ChannelManager::StartAecDump(rtc::PlatformFile file) {
+ return worker_thread_->Invoke<bool>(
+ Bind(&MediaEngineInterface::StartAecDump, media_engine_.get(), file));
}
void ChannelManager::StopAecDump() {
diff --git a/talk/session/media/channelmanager.h b/talk/session/media/channelmanager.h
index f69bf9a..2bc516b 100644
--- a/talk/session/media/channelmanager.h
+++ b/talk/session/media/channelmanager.h
@@ -162,10 +162,8 @@
// The operations below occur on the main thread.
- // Starts AEC dump using existing file, with a specified maximum file size in
- // bytes. When the limit is reached, logging will stop and the file will be
- // closed. If max_size_bytes is set to <= 0, no limit will be used.
- bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
+ // Starts AEC dump using existing file.
+ bool StartAecDump(rtc::PlatformFile file);
// Stops recording AEC dump.
void StopAecDump();
diff --git a/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
index ce607e9..523c0e0 100644
--- a/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
+++ b/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -496,7 +496,7 @@
ParcelFileDescriptor.MODE_READ_WRITE |
ParcelFileDescriptor.MODE_CREATE |
ParcelFileDescriptor.MODE_TRUNCATE);
- factory.startAecDump(aecDumpFileDescriptor.getFd(), -1);
+ factory.startAecDump(aecDumpFileDescriptor.getFd());
} catch(IOException e) {
Log.e(TAG, "Can not open aecdump file", e);
}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index b79b4f0..a332945 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -632,7 +632,6 @@
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
@@ -720,7 +719,6 @@
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
@@ -888,7 +886,6 @@
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
@@ -957,7 +954,6 @@
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
@@ -1043,8 +1039,7 @@
}
int AudioProcessingImpl::StartDebugRecording(
- const char filename[AudioProcessing::kMaxFilenameSize],
- int64_t max_log_size_bytes) {
+ const char filename[AudioProcessing::kMaxFilenameSize]) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
@@ -1055,7 +1050,6 @@
}
#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) {
@@ -1076,8 +1070,7 @@
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
-int AudioProcessingImpl::StartDebugRecording(FILE* handle,
- int64_t max_log_size_bytes) {
+int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
// Run in a single-threaded manner.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
@@ -1087,8 +1080,6 @@
}
#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) {
@@ -1114,7 +1105,7 @@
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
- return StartDebugRecording(stream, -1);
+ return StartDebugRecording(stream);
}
int AudioProcessingImpl::StopDebugRecording() {
@@ -1409,7 +1400,6 @@
#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();
@@ -1427,19 +1417,7 @@
{
// Ensure atomic writes of the message.
- 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;
- }
- }
+ rtc::CritScope cs_capture(crit_debug);
// Write message preceded by its size.
if (!debug_file->Write(&size, sizeof(int32_t))) {
return kFileError;
@@ -1474,7 +1452,6 @@
// 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;
}
@@ -1527,7 +1504,6 @@
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;
}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index b720bbf..3506ac4 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -57,9 +57,8 @@
int Initialize(const ProcessingConfig& processing_config) override;
void SetExtraOptions(const Config& config) override;
void UpdateHistogramsOnCallEnd() override;
- int StartDebugRecording(const char filename[kMaxFilenameSize],
- int64_t max_log_size_bytes) override;
- int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override;
+ int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
+ int StartDebugRecording(FILE* handle) override;
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
int StopDebugRecording() override;
@@ -144,9 +143,6 @@
struct ApmDebugDumpState {
ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
- // Number of bytes that can still be written to the log before the maximum
- // size is reached. A value of <= 0 indicates that no limit is used.
- int64_t num_bytes_left_for_log_ = -1;
rtc::scoped_ptr<FileWrapper> debug_file;
ApmDebugDumpThreadState render;
ApmDebugDumpThreadState capture;
@@ -225,7 +221,6 @@
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
static int WriteMessageToDebugFile(FileWrapper* debug_file,
- int64_t* filesize_limit_bytes,
rtc::CriticalSection* crit_debug,
ApmDebugDumpThreadState* debug_state);
int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index c96f23a..5fcc4d4 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -408,22 +408,13 @@
// Starts recording debugging information to a file specified by |filename|,
// a NULL-terminated string. If there is an ongoing recording, the old file
// will be closed, and recording will continue in the newly specified file.
- // An already existing file will be overwritten without warning. A maximum
- // file size (in bytes) for the log can be specified. The logging is stopped
- // once the limit has been reached. If max_log_size_bytes is set to a value
- // <= 0, no limit will be used.
+ // An already existing file will be overwritten without warning.
static const size_t kMaxFilenameSize = 1024;
- virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
- int64_t max_log_size_bytes) = 0;
+ virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
// Same as above but uses an existing file handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
- virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
-
- // TODO(ivoc): Remove this function after Chromium switches to the one above.
- virtual int StartDebugRecording(FILE* handle) {
- return StartDebugRecording(handle, -1);
- }
+ virtual int StartDebugRecording(FILE* handle) = 0;
// Same as above but uses an existing PlatformFile handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
index 3aea406..4ff52ba 100644
--- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -250,11 +250,10 @@
void(int offset));
MOCK_CONST_METHOD0(delay_offset_ms,
int());
- MOCK_METHOD2(StartDebugRecording,
- int(const char filename[kMaxFilenameSize],
- int64_t max_log_size_bytes));
- MOCK_METHOD2(StartDebugRecording,
- int(FILE* handle, int64_t max_log_size_bytes));
+ MOCK_METHOD1(StartDebugRecording,
+ int(const char filename[kMaxFilenameSize]));
+ MOCK_METHOD1(StartDebugRecording,
+ int(FILE* handle));
MOCK_METHOD0(StopDebugRecording,
int());
MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index b6b94f0..eff791d 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -388,8 +388,7 @@
int AnalyzeReverseStreamChooser(Format format);
void ProcessDebugDump(const std::string& in_filename,
const std::string& out_filename,
- Format format,
- int max_size_bytes);
+ Format format);
void VerifyDebugDumpTest(Format format);
const std::string output_path_;
@@ -1712,8 +1711,7 @@
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
void ApmTest::ProcessDebugDump(const std::string& in_filename,
const std::string& out_filename,
- Format format,
- int max_size_bytes) {
+ Format format) {
FILE* in_file = fopen(in_filename.c_str(), "rb");
ASSERT_TRUE(in_file != NULL);
audioproc::Event event_msg;
@@ -1741,8 +1739,7 @@
if (first_init) {
// StartDebugRecording() writes an additional init message. Don't start
// recording until after the first init to avoid the extra message.
- EXPECT_NOERR(
- apm_->StartDebugRecording(out_filename.c_str(), max_size_bytes));
+ EXPECT_NOERR(apm_->StartDebugRecording(out_filename.c_str()));
first_init = false;
}
@@ -1815,54 +1812,34 @@
test::OutputPath(), std::string("ref") + format_string + "_aecdump");
const std::string out_filename = test::TempFilename(
test::OutputPath(), std::string("out") + format_string + "_aecdump");
- const std::string limited_filename = test::TempFilename(
- test::OutputPath(), std::string("limited") + format_string + "_aecdump");
- const size_t logging_limit_bytes = 100000;
- // We expect at least this many bytes in the created logfile.
- const size_t logging_expected_bytes = 95000;
EnableAllComponents();
- ProcessDebugDump(in_filename, ref_filename, format, -1);
- ProcessDebugDump(ref_filename, out_filename, format, -1);
- ProcessDebugDump(ref_filename, limited_filename, format, logging_limit_bytes);
+ ProcessDebugDump(in_filename, ref_filename, format);
+ ProcessDebugDump(ref_filename, out_filename, format);
FILE* ref_file = fopen(ref_filename.c_str(), "rb");
FILE* out_file = fopen(out_filename.c_str(), "rb");
- FILE* limited_file = fopen(limited_filename.c_str(), "rb");
ASSERT_TRUE(ref_file != NULL);
ASSERT_TRUE(out_file != NULL);
- ASSERT_TRUE(limited_file != NULL);
rtc::scoped_ptr<uint8_t[]> ref_bytes;
rtc::scoped_ptr<uint8_t[]> out_bytes;
- rtc::scoped_ptr<uint8_t[]> limited_bytes;
size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
size_t out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
- size_t limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
size_t bytes_read = 0;
- size_t bytes_read_limited = 0;
while (ref_size > 0 && out_size > 0) {
bytes_read += ref_size;
- bytes_read_limited += limited_size;
EXPECT_EQ(ref_size, out_size);
- EXPECT_GE(ref_size, limited_size);
EXPECT_EQ(0, memcmp(ref_bytes.get(), out_bytes.get(), ref_size));
- EXPECT_EQ(0, memcmp(ref_bytes.get(), limited_bytes.get(), limited_size));
ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
- limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
}
EXPECT_GT(bytes_read, 0u);
- EXPECT_GT(bytes_read_limited, logging_expected_bytes);
- EXPECT_LE(bytes_read_limited, logging_limit_bytes);
EXPECT_NE(0, feof(ref_file));
EXPECT_NE(0, feof(out_file));
- EXPECT_NE(0, feof(limited_file));
ASSERT_EQ(0, fclose(ref_file));
ASSERT_EQ(0, fclose(out_file));
- ASSERT_EQ(0, fclose(limited_file));
remove(ref_filename.c_str());
remove(out_filename.c_str());
- remove(limited_filename.c_str());
}
TEST_F(ApmTest, VerifyDebugDumpInt) {
@@ -1879,13 +1856,13 @@
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
EXPECT_EQ(apm_->kNullPointerError,
- apm_->StartDebugRecording(static_cast<const char*>(NULL), -1));
+ apm_->StartDebugRecording(static_cast<const char*>(NULL)));
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str(), -1));
+ EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str()));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
@@ -1899,7 +1876,7 @@
ASSERT_EQ(0, remove(filename.c_str()));
#else
EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(filename.c_str(), -1));
+ apm_->StartDebugRecording(filename.c_str()));
EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
// Verify the file has NOT been written.
@@ -1910,7 +1887,7 @@
// TODO(andrew): expand test to verify output.
TEST_F(ApmTest, DebugDumpFromFileHandle) {
FILE* fid = NULL;
- EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid, -1));
+ EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid));
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
fid = fopen(filename.c_str(), "w");
@@ -1920,7 +1897,7 @@
// Stopping without having started should be OK.
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid, -1));
+ EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid));
EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
@@ -1934,7 +1911,7 @@
ASSERT_EQ(0, remove(filename.c_str()));
#else
EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(fid, -1));
+ apm_->StartDebugRecording(fid));
EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
ASSERT_EQ(0, fclose(fid));
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
index 6aa310c..d2dd9c8 100644
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -181,7 +181,7 @@
}
void DebugDumpGenerator::StartRecording() {
- apm_->StartDebugRecording(dump_file_name_.c_str(), -1);
+ apm_->StartDebugRecording(dump_file_name_.c_str());
}
void DebugDumpGenerator::Process(size_t num_blocks) {
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index f54eab3..ae6b4dc 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -434,7 +434,7 @@
} else if (strcmp(argv[i], "--debug_file") == 0) {
i++;
ASSERT_LT(i, argc) << "Specify filename after --debug_file";
- ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i], -1));
+ ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i]));
} else {
FAIL() << "Unrecognized argument " << argv[i];
}
diff --git a/webrtc/voice_engine/voe_audio_processing_impl.cc b/webrtc/voice_engine/voe_audio_processing_impl.cc
index 4479fe0..c957263 100644
--- a/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -924,7 +924,7 @@
return -1;
}
- return _shared->audio_processing()->StartDebugRecording(fileNameUTF8, -1);
+ return _shared->audio_processing()->StartDebugRecording(fileNameUTF8);
}
int VoEAudioProcessingImpl::StartDebugRecording(FILE* file_handle) {
@@ -935,7 +935,7 @@
return -1;
}
- return _shared->audio_processing()->StartDebugRecording(file_handle, -1);
+ return _shared->audio_processing()->StartDebugRecording(file_handle);
}
int VoEAudioProcessingImpl::StopDebugRecording() {
