Remove traces of deprecated WebRtc_Word types.
BUG=314
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1385004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index fd0d1d9..5001da2 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -266,23 +266,23 @@
struct NetworkStatistics // NETEQ statistics
{
// current jitter buffer size in ms
- WebRtc_UWord16 currentBufferSize;
+ uint16_t currentBufferSize;
// preferred (optimal) buffer size in ms
- WebRtc_UWord16 preferredBufferSize;
+ uint16_t preferredBufferSize;
// adding extra delay due to "peaky jitter"
bool jitterPeaksFound;
// loss rate (network + late) in percent (in Q14)
- WebRtc_UWord16 currentPacketLossRate;
+ uint16_t currentPacketLossRate;
// late loss rate in percent (in Q14)
- WebRtc_UWord16 currentDiscardRate;
+ uint16_t currentDiscardRate;
// fraction (of original stream) of synthesized speech inserted through
// expansion (in Q14)
- WebRtc_UWord16 currentExpandRate;
+ uint16_t currentExpandRate;
// fraction of synthesized speech inserted through pre-emptive expansion
// (in Q14)
- WebRtc_UWord16 currentPreemptiveRate;
+ uint16_t currentPreemptiveRate;
// fraction of data removed through acceleration (in Q14)
- WebRtc_UWord16 currentAccelerateRate;
+ uint16_t currentAccelerateRate;
// clock-drift in parts-per-million (negative or positive)
int32_t clockDriftPPM;
// average packet waiting time in the jitter buffer (ms)
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 2d02062..0a36a9e 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -82,8 +82,7 @@
MOCK_METHOD1(SetRtxReceivePayloadType,
void(int));
MOCK_METHOD2(IncomingPacket,
- int32_t(const WebRtc_UWord8* incomingPacket,
- const WebRtc_UWord16 packetLength));
+ int32_t(const uint8_t* incomingPacket, const uint16_t packetLength));
MOCK_METHOD4(IncomingAudioNTP,
int32_t(const uint32_t audioReceivedNTPsecs,
const uint32_t audioReceivedNTPfrac,
diff --git a/webrtc/modules/video_coding/main/test/receiver_tests.h b/webrtc/modules/video_coding/main/test/receiver_tests.h
index 15ec424..095a231 100644
--- a/webrtc/modules/video_coding/main/test/receiver_tests.h
+++ b/webrtc/modules/video_coding/main/test/receiver_tests.h
@@ -26,9 +26,9 @@
RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
virtual ~RtpDataCallback() {}
- virtual WebRtc_Word32 OnReceivedPayloadData(
- const WebRtc_UWord8* payload_data,
- const WebRtc_UWord16 payload_size,
+ virtual int32_t OnReceivedPayloadData(
+ const uint8_t* payload_data,
+ const uint16_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) {
return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
}
diff --git a/webrtc/modules/video_coding/main/test/test_util.cc b/webrtc/modules/video_coding/main/test/test_util.cc
index c16b98d..159de17 100644
--- a/webrtc/modules/video_coding/main/test/test_util.cc
+++ b/webrtc/modules/video_coding/main/test/test_util.cc
@@ -111,7 +111,7 @@
}
}
-WebRtc_Word32 FileOutputFrameReceiver::FrameToRender(
+int32_t FileOutputFrameReceiver::FrameToRender(
webrtc::I420VideoFrame& video_frame) {
if (timing_file_ == NULL) {
std::string basename;
diff --git a/webrtc/modules/video_coding/main/test/test_util.h b/webrtc/modules/video_coding/main/test/test_util.h
index cfbd24b..43af6f1 100644
--- a/webrtc/modules/video_coding/main/test/test_util.h
+++ b/webrtc/modules/video_coding/main/test/test_util.h
@@ -87,7 +87,7 @@
virtual ~FileOutputFrameReceiver();
// VCMReceiveCallback
- virtual WebRtc_Word32 FrameToRender(webrtc::I420VideoFrame& video_frame);
+ virtual int32_t FrameToRender(webrtc::I420VideoFrame& video_frame);
private:
std::string out_filename_;
diff --git a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
index 0026767..a84561a 100644
--- a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
+++ b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
@@ -60,21 +60,22 @@
}
// PayloadSinkInterface
- virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payload_data,
- const WebRtc_UWord16 payload_size,
+ virtual int32_t OnReceivedPayloadData(
+ const uint8_t* payload_data,
+ const uint16_t payload_size,
const WebRtcRTPHeader* rtp_header) {
return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
}
// VCMPacketRequestCallback
- virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequence_numbers,
- WebRtc_UWord16 length) {
+ virtual int32_t ResendPackets(const uint16_t* sequence_numbers,
+ uint16_t length) {
stream_->ResendPackets(sequence_numbers, length);
return 0;
}
// VCMFrameStorageCallback
- virtual WebRtc_Word32 StoreReceivedFrame(
+ virtual int32_t StoreReceivedFrame(
const EncodedVideoData& frame_to_store) {
vcm_playback_->DecodeFromStorage(frame_to_store);
return VCM_OK;
diff --git a/webrtc/typedefs.h b/webrtc/typedefs.h
index 0d2dfcd..7edb12c 100644
--- a/webrtc/typedefs.h
+++ b/webrtc/typedefs.h
@@ -75,17 +75,6 @@
typedef unsigned __int64 uint64_t;
#endif
-// TODO(andrew): remove WebRtc_ types:
-// http://code.google.com/p/webrtc/issues/detail?id=314
-typedef int8_t WebRtc_Word8;
-typedef int16_t WebRtc_Word16;
-typedef int32_t WebRtc_Word32;
-typedef int64_t WebRtc_Word64;
-typedef uint8_t WebRtc_UWord8;
-typedef uint16_t WebRtc_UWord16;
-typedef uint32_t WebRtc_UWord32;
-typedef uint64_t WebRtc_UWord64;
-
// Borrowed from Chromium's base/compiler_specific.h.
// Annotate a virtual method indicating it must be overriding a virtual
// method in the parent class.
