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The AudioProcessing class is used as an interface

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to the functionality in the audio processing module.
Therefore, it should be a pure interface.
This CL ensures that is the case.

BUG=webrtc:6515

Review-Url: https://codereview.webrtc.org/2406193002
Cr-Commit-Position: refs/heads/master@{#14608}
This commit is contained in:
peah
2016-10-12 03:01:49 -07:00
committed by Commit bot
parent 4b8bfb8ed3
commit 73a28ee066
6 changed files with 13 additions and 9 deletions
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2
webrtc/media/engine/fakewebrtcvoiceengine.h
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@ -116,6 +116,8 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
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, (FILE * handle));
WEBRTC_STUB(StartDebugRecordingForPlatformFile, (rtc::PlatformFile handle));
WEBRTC_STUB(StopDebugRecording, ());
WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }

4
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -1251,6 +1251,10 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle,
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
return StartDebugRecording(handle, -1);
}
int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
rtc::PlatformFile handle) {
// Run in a single-threaded manner.

2
webrtc/modules/audio_processing/audio_processing_impl.h
View File

@ -65,7 +65,7 @@ class AudioProcessingImpl : public AudioProcessing {
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;

4
webrtc/modules/audio_processing/include/audio_processing.cc
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@ -31,9 +31,5 @@ Beamforming::Beamforming(bool enabled,
Beamforming::~Beamforming() {}
int AudioProcessing::StartDebugRecordingForPlatformFile(
rtc::PlatformFile handle) {
return -1;
}
} // namespace webrtc

6
webrtc/modules/audio_processing/include/audio_processing.h
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@ -454,14 +454,12 @@ class AudioProcessing {
virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
// TODO(ivoc): Remove this function after Chrome stops using it.
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().
// TODO(xians): Make this interface pure virtual.
virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle);
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).

4
webrtc/modules/audio_processing/include/mock_audio_processing.h
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@ -259,6 +259,10 @@ class MockAudioProcessing : public AudioProcessing {
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(StartDebugRecording,
int(rtc::PlatformFile handle));
MOCK_METHOD0(StopDebugRecording,
int());
MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());