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Remove older AEC-dump interface.

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This CL completely removes the methods
AudioProcessing::{Start,Stop}DebugDumpRecording. These methods have
been replaced with AudioProcessing::{Attach,Detach}AecDump. Their
implementation was removed in the parent CL
https://chromium-review.googlesource.com/c/589147

Bug: webrtc:7404
Change-Id: Ia3d5314985af9c74f79c94c514ded1f8afc78fb5
Reviewed-on: https://chromium-review.googlesource.com/589152
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19334}
This commit is contained in:
Alex Loiko
2017-07-27 13:10:23 +02:00
committed by Commit Bot
parent f3238e43ce
commit dc5fc82c62
4 changed files with 0 additions and 71 deletions
Download Patch File Download Diff File Expand all files Collapse all files

30
webrtc/modules/audio_processing/audio_processing_impl.cc
View File

@ -1517,36 +1517,6 @@ void AudioProcessingImpl::DetachAecDump() {
}
}
int AudioProcessingImpl::StartDebugRecording(
const char filename[AudioProcessing::kMaxFilenameSize],
int64_t max_log_size_bytes) {
RTC_NOTREACHED();
return kUnsupportedFunctionError;
}
int AudioProcessingImpl::StartDebugRecording(FILE* handle,
int64_t max_log_size_bytes) {
RTC_NOTREACHED();
return kUnsupportedFunctionError;
}
int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
RTC_NOTREACHED();
return kUnsupportedFunctionError;
}
int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
rtc::PlatformFile handle) {
RTC_NOTREACHED();
return kUnsupportedFunctionError;
}
int AudioProcessingImpl::StopDebugRecording() {
// DetachAecDump();
RTC_NOTREACHED();
return kUnsupportedFunctionError;
}
AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);

6
webrtc/modules/audio_processing/audio_processing_impl.h
View File

@ -56,12 +56,6 @@ class AudioProcessingImpl : public AudioProcessing {
void UpdateHistogramsOnCallEnd() override;
void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
void DetachAecDump() 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(FILE* handle) override;
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
int StopDebugRecording() override;
// Capture-side exclusive methods possibly running APM in a
// multi-threaded manner. Acquire the capture lock.

27
webrtc/modules/audio_processing/include/audio_processing.h
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@ -471,33 +471,6 @@ class AudioProcessing : public rtc::RefCountInterface {
// all pending logging tasks are completed.
virtual void DetachAecDump() = 0;
// 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.
static const size_t kMaxFilenameSize = 1024;
virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
int64_t max_log_size_bytes) = 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 Chrome stops using it.
virtual int StartDebugRecording(FILE* handle) = 0;
// Same as above but uses an existing PlatformFile handle. Takes ownership
// of |handle| and closes it at StopDebugRecording().
// TODO(xians): Make this interface pure virtual.
virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) = 0;
// Stops recording debugging information, and closes the file. Recording
// cannot be resumed in the same file (without overwriting it).
virtual int StopDebugRecording() = 0;
// Use to send UMA histograms at end of a call. Note that all histogram
// specific member variables are reset.
virtual void UpdateHistogramsOnCallEnd() = 0;

8
webrtc/modules/audio_processing/include/mock_audio_processing.h
View File

@ -179,14 +179,6 @@ class MockAudioProcessing : public AudioProcessing {
virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {}
MOCK_METHOD0(DetachAecDump, void());
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 (FILE* handle));
MOCK_METHOD1(StartDebugRecordingForPlatformFile,
int(rtc::PlatformFile handle));
MOCK_METHOD0(StopDebugRecording, int());
MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
virtual MockEchoCancellation* echo_cancellation() const {