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Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."

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> Revert 3933 "Remove traces of deprecated WebRtc_Word types."
> 
> > Remove traces of deprecated WebRtc_Word types.
> > 
> > BUG=314
> > R=tommi@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/1385004
> 
> TBR=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1386004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1397004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-05-03 12:02:11 +00:00
parent 2580bc4c30
commit 77f6b2175e
7 changed files with 19 additions and 30 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

14
webrtc/common_types.h
View File

@ -266,23 +266,23 @@ enum VadModes // degree of bandwidth reduction
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)

3
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
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@ -82,8 +82,7 @@ class MockRtpRtcp : public RtpRtcp {
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,

6
webrtc/modules/video_coding/main/test/receiver_tests.h
View File

@ -26,9 +26,9 @@ class RtpDataCallback : public webrtc::RtpData {
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);
}

2
webrtc/modules/video_coding/main/test/test_util.cc
View File

@ -111,7 +111,7 @@ FileOutputFrameReceiver::~FileOutputFrameReceiver() {
}
}
WebRtc_Word32 FileOutputFrameReceiver::FrameToRender(
int32_t FileOutputFrameReceiver::FrameToRender(
webrtc::I420VideoFrame& video_frame) {
if (timing_file_ == NULL) {
std::string basename;

2
webrtc/modules/video_coding/main/test/test_util.h
View File

@ -87,7 +87,7 @@ class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
virtual ~FileOutputFrameReceiver();
// VCMReceiveCallback
virtual WebRtc_Word32 FrameToRender(webrtc::I420VideoFrame& video_frame);
virtual int32_t FrameToRender(webrtc::I420VideoFrame& video_frame);
private:
std::string out_filename_;

11
webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
View File

@ -60,21 +60,22 @@ class VcmPayloadSinkFactory::VcmPayloadSink
}
// 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;

11
webrtc/typedefs.h
View File

@ -75,17 +75,6 @@ typedef unsigned int uint32_t;
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.