From 7aad5e5cced724c08b0af4bf85db446c5965ac76 Mon Sep 17 00:00:00 2001 From: "xians@webrtc.org" Date: Tue, 30 Sep 2014 15:26:15 +0000 Subject: [PATCH] Revert 7338 "Fixed the android build by making the interface pur..." > Fixed the android build by making the interface pure virtual. > > TBR=asapersson@webrtc.org, bjornv@webrtc.org, > > Review URL: https://webrtc-codereview.appspot.com/24789004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d --- talk/media/webrtc/fakewebrtcvoiceengine.h | 2 -- webrtc/modules/audio_processing/include/audio_processing.h | 2 +- webrtc/modules/audio_processing/include/mock_audio_processing.h | 2 -- 3 files changed, 1 insertion(+), 5 deletions(-) diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h index 542581a8f3..52a50ff284 100644 --- a/talk/media/webrtc/fakewebrtcvoiceengine.h +++ b/talk/media/webrtc/fakewebrtcvoiceengine.h @@ -38,7 +38,6 @@ #include "talk/media/webrtc/fakewebrtccommon.h" #include "talk/media/webrtc/webrtcvoe.h" #include "webrtc/base/basictypes.h" -#include "webrtc/base/fileutils.h" #include "webrtc/base/gunit.h" #include "webrtc/base/stringutils.h" #ifdef USE_WEBRTC_DEV_BRANCH @@ -129,7 +128,6 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { WEBRTC_STUB_CONST(delay_offset_ms, ()); WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize])); WEBRTC_STUB(StartDebugRecording, (FILE* handle)); - WEBRTC_STUB(StartDebugRecording, (rtc::PlatformFile handle)); WEBRTC_STUB(StopDebugRecording, ()); virtual webrtc::EchoCancellation* echo_cancellation() const OVERRIDE { return NULL; diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h index bcc83ae470..53157ae66f 100644 --- a/webrtc/modules/audio_processing/include/audio_processing.h +++ b/webrtc/modules/audio_processing/include/audio_processing.h @@ -329,7 +329,7 @@ class AudioProcessing { // 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 StartDebugRecording(rtc::PlatformFile handle) = 0; + virtual int StartDebugRecording(rtc::PlatformFile handle) { return -1; } // Stops recording debugging information, and closes the file. Recording // cannot be resumed in the same file (without overwriting it). diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h index 3f2b1f9510..8258bb6cff 100644 --- a/webrtc/modules/audio_processing/include/mock_audio_processing.h +++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h @@ -239,8 +239,6 @@ class MockAudioProcessing : public AudioProcessing { int(const char filename[kMaxFilenameSize])); MOCK_METHOD1(StartDebugRecording, int(FILE* handle)); - MOCK_METHOD1(StartDebugRecording, - int(rtc::PlatformFile handle)); MOCK_METHOD0(StopDebugRecording, int()); virtual MockEchoCancellation* echo_cancellation() const {