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pre-factor cleanup pre-work.

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Review URL: https://webrtc-codereview.appspot.com/938010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-11-07 17:01:04 +00:00
parent 4cebe6cded
commit c66e8b3f31
4 changed files with 803 additions and 1111 deletions
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34
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -52,7 +52,11 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: _rtpSender(configuration.id, configuration.audio, configuration.clock),
: _rtpSender(configuration.id,
configuration.audio,
configuration.clock,
configuration.outgoing_transport,
configuration.audio_messages),
_rtpReceiver(configuration.id, configuration.audio, configuration.clock,
this),
_rtcpSender(configuration.id, configuration.audio, configuration.clock,
@ -97,10 +101,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
_rtcpReceiver.RegisterRtcpObservers(configuration.intra_frame_callback,
configuration.bandwidth_callback,
configuration.rtcp_feedback);
_rtpSender.RegisterAudioCallback(configuration.audio_messages);
_rtpReceiver.RegisterIncomingAudioCallback(configuration.audio_messages);
_rtpSender.RegisterSendTransport(configuration.outgoing_transport);
_rtcpSender.RegisterSendTransport(configuration.outgoing_transport);
// make sure that RTCP objects are aware of our SSRC
@ -628,7 +630,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStartTimestamp(
"SetStartTimestamp(%d)",
timestamp);
_rtcpSender.SetStartTimestamp(timestamp);
return _rtpSender.SetStartTimestamp(timestamp, true);
_rtpSender.SetStartTimestamp(timestamp, true);
return 0; // TODO(pwestin): change to void.
}
WebRtc_UWord16 ModuleRtpRtcpImpl::SequenceNumber() const {
@ -646,7 +649,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetSequenceNumber(
"SetSequenceNumber(%d)",
seqNum);
return _rtpSender.SetSequenceNumber(seqNum);
_rtpSender.SetSequenceNumber(seqNum);
return 0; // TODO(pwestin): change to void.
}
WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
@ -659,17 +663,16 @@ WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRC(const WebRtc_UWord32 ssrc) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetSSRC(%d)", ssrc);
if (_rtpSender.SetSSRC(ssrc) == 0) {
_rtpSender.SetSSRC(ssrc);
_rtcpReceiver.SetSSRC(ssrc);
_rtcpSender.SetSSRC(ssrc);
return 0;
}
return -1;
return 0; // TODO(pwestin): change to void.
}
WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) {
_rtcpSender.SetCSRCStatus(include);
return _rtpSender.SetCSRCStatus(include);
_rtpSender.SetCSRCStatus(include);
return 0; // TODO(pwestin): change to void.
}
WebRtc_Word32 ModuleRtpRtcpImpl::CSRCs(
@ -702,16 +705,15 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCs(
}
it++;
}
return 0;
} else {
for (int i = 0; i < arrLength; i++) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "\tidx:%d CSRC:%u", i,
arrOfCSRC[i]);
}
_rtcpSender.SetCSRCs(arrOfCSRC, arrLength);
return _rtpSender.SetCSRCs(arrOfCSRC, arrLength);
_rtpSender.SetCSRCs(arrOfCSRC, arrLength);
}
return 0; // TODO(pwestin): change to void.
}
WebRtc_UWord32 ModuleRtpRtcpImpl::PacketCountSent() const {
@ -1129,7 +1131,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
"ResetSendDataCountersRTP()");
return _rtpSender.ResetDataCounters();
_rtpSender.ResetDataCounters();
return 0; // TODO(pwestin): change to void.
}
// Force a send of an RTCP packet
@ -1495,7 +1498,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus(
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
"SetStorePacketsStatus(disable)");
}
return _rtpSender.SetStorePacketsStatus(enable, numberToStore);
_rtpSender.SetStorePacketsStatus(enable, numberToStore);
return 0; // TODO(pwestin): change to void.
}
/*

642
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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File diff suppressed because it is too large Load Diff

62
webrtc/modules/rtp_rtcp/source/rtp_sender.h
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@ -17,7 +17,7 @@
#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
#include "rtp_rtcp_defines.h"
#include "common_types.h" // Encryption
#include "common_types.h"
#include "ssrc_database.h"
#include "Bitrate.h"
#include "rtp_header_extension.h"
@ -32,9 +32,8 @@ class RTPPacketHistory;
class RTPSenderAudio;
class RTPSenderVideo;
class RTPSenderInterface
{
public:
class RTPSenderInterface {
public:
RTPSenderInterface() {}
virtual ~RTPSenderInterface() {}
@ -57,16 +56,16 @@ public:
virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* data_buffer,
const WebRtc_UWord16 payload_length,
const WebRtc_UWord16 rtp_header_length,
WebRtc_UWord16 payload_length,
WebRtc_UWord16 rtp_header_length,
int64_t capture_time_ms,
const StorageType storage) = 0;
StorageType storage) = 0;
};
class RTPSender : public Bitrate, public RTPSenderInterface
{
public:
RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock);
class RTPSender : public Bitrate, public RTPSenderInterface {
public:
RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock,
Transport* transport, RtpAudioFeedback* audio_feedback);
virtual ~RTPSender();
void ProcessBitrate();
@ -82,9 +81,6 @@ public:
WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
// callback
WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
WebRtc_Word32 RegisterPayload(
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType,
@ -109,23 +105,22 @@ public:
// number of sent RTP bytes
WebRtc_UWord32 Bytes() const;
WebRtc_Word32 ResetDataCounters();
void ResetDataCounters();
WebRtc_UWord32 StartTimestamp() const;
WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp,
const bool force = false);
void SetStartTimestamp(WebRtc_UWord32 timestamp, bool force);
WebRtc_UWord32 GenerateNewSSRC();
WebRtc_Word32 SetSSRC( const WebRtc_UWord32 ssrc);
void SetSSRC(const WebRtc_UWord32 ssrc);
WebRtc_UWord16 SequenceNumber() const;
WebRtc_Word32 SetSequenceNumber( WebRtc_UWord16 seq);
void SetSequenceNumber(WebRtc_UWord16 seq);
WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
WebRtc_Word32 SetCSRCStatus(const bool include);
void SetCSRCStatus(const bool include);
WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
void SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
const WebRtc_UWord8 arrLength);
WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
@ -181,7 +176,7 @@ public:
const WebRtc_UWord16* nackSequenceNumbers,
const WebRtc_UWord16 avgRTT);
WebRtc_Word32 SetStorePacketsStatus(const bool enable,
void SetStorePacketsStatus(const bool enable,
const WebRtc_UWord16 numberToStore);
bool StorePackets() const;
@ -223,16 +218,13 @@ public:
virtual WebRtc_UWord32 SSRC() const;
virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* data_buffer,
const WebRtc_UWord16 payload_length,
const WebRtc_UWord16 rtp_header_length,
WebRtc_UWord16 payload_length,
WebRtc_UWord16 rtp_header_length,
int64_t capture_time_ms,
const StorageType storage);
StorageType storage);
/*
* Audio
*/
WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
// Send a DTMF tone using RFC 2833 (4733)
WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
const WebRtc_UWord16 time_ms,
@ -240,7 +232,8 @@ public:
bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
// Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG)
WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
// Set status and ID for header-extension-for-audio-level-indication.
@ -251,7 +244,8 @@ public:
WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
WebRtc_UWord8& ID) const;
// Store the audio level in dBov for header-extension-for-audio-level-indication.
// Store the audio level in dBov for
// header-extension-for-audio-level-indication.
WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
// Set payload type for Redundant Audio Data RFC 2198
@ -284,11 +278,11 @@ public:
const FecProtectionParams* delta_params,
const FecProtectionParams* key_params);
protected:
protected:
WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
RtpVideoCodecTypes& videoType);
private:
private:
void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
const WebRtc_UWord32 now);
@ -304,9 +298,7 @@ private:
CriticalSectionWrapper* _sendCritsect;
CriticalSectionWrapper* _transportCritsect;
Transport* _transport;
bool _sendingMedia;
WebRtc_UWord16 _maxPayloadLength;
@ -329,7 +321,7 @@ private:
WebRtc_Word64 _timeLastSendToNetworkUpdate;
bool _transmissionSmoothing;
// statistics
// Statistics
WebRtc_UWord32 _packetsSent;
WebRtc_UWord32 _payloadBytesSent;

10
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -78,12 +78,11 @@ class RtpSenderTest : public ::testing::Test {
protected:
RtpSenderTest()
: fake_clock_(),
rtp_sender_(new RTPSender(0, false, &fake_clock_)),
transport_(),
rtp_sender_(new RTPSender(0, false, &fake_clock_, &transport_, NULL)),
kMarkerBit(true),
kType(kRtpExtensionTransmissionTimeOffset),
packet_() {
EXPECT_EQ(0, rtp_sender_->SetSequenceNumber(kSeqNum));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
~RtpSenderTest() {
delete rtp_sender_;
@ -199,8 +198,6 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
}
TEST_F(RtpSenderTest, NoTrafficSmoothing) {
EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
@ -218,9 +215,8 @@ TEST_F(RtpSenderTest, NoTrafficSmoothing) {
TEST_F(RtpSenderTest, TrafficSmoothing) {
rtp_sender_->SetTransmissionSmoothingStatus(true);
EXPECT_EQ(0, rtp_sender_->SetStorePacketsStatus(true, 10));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
rtp_sender_->SetTargetSendBitrate(300000);
WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,