Index: talk/app/webrtc/test/fakeaudiocapturemodule.h |
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/talk/app/webrtc/test/fakeaudiocapturemodule.h |
index 8ff4aa19e7552eea7a0520bf6f4907ce865a081c..be1df9a7bd7e48c249b8011256994d9757bb7cc7 100644 |
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.h |
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.h |
@@ -40,14 +40,13 @@ |
#include "webrtc/base/basictypes.h" |
#include "webrtc/base/criticalsection.h" |
#include "webrtc/base/messagehandler.h" |
+#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/base/scoped_ref_ptr.h" |
#include "webrtc/common_types.h" |
#include "webrtc/modules/audio_device/include/audio_device.h" |
namespace rtc { |
- |
class Thread; |
- |
} // namespace rtc |
class FakeAudioCaptureModule |
@@ -62,10 +61,7 @@ class FakeAudioCaptureModule |
static const int kNumberBytesPerSample = sizeof(Sample); |
// Creates a FakeAudioCaptureModule or returns NULL on failure. |
- // |process_thread| is used to push and pull audio frames to and from the |
- // returned instance. Note: ownership of |process_thread| is not handed over. |
- static rtc::scoped_refptr<FakeAudioCaptureModule> Create( |
- rtc::Thread* process_thread); |
+ static rtc::scoped_refptr<FakeAudioCaptureModule> Create(); |
// Returns the number of frames that have been successfully pulled by the |
// instance. Note that correctly detecting success can only be done if the |
@@ -206,7 +202,7 @@ class FakeAudioCaptureModule |
// exposed in which case the burden of proper instantiation would be put on |
// the creator of a FakeAudioCaptureModule instance. To create an instance of |
// this class use the Create(..) API. |
- explicit FakeAudioCaptureModule(rtc::Thread* process_thread); |
+ explicit FakeAudioCaptureModule(); |
// The destructor is protected because it is reference counted and should not |
// be deleted directly. |
virtual ~FakeAudioCaptureModule(); |
@@ -239,8 +235,6 @@ class FakeAudioCaptureModule |
void ReceiveFrameP(); |
// Pushes frames to the registered webrtc::AudioTransport. |
void SendFrameP(); |
- // Stops the periodic calling of ProcessFrame() in a thread safe way. |
- void StopProcessP(); |
// The time in milliseconds when Process() was last called or 0 if no call |
// has been made. |
@@ -266,8 +260,7 @@ class FakeAudioCaptureModule |
bool started_; |
uint32 next_frame_time_; |
- // User provided thread context. |
- rtc::Thread* process_thread_; |
+ rtc::scoped_ptr<rtc::Thread> process_thread_; |
// Buffer for storing samples received from the webrtc::AudioTransport. |
char rec_buffer_[kNumberSamples * kNumberBytesPerSample]; |