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RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies.

Browse Source 
BUG=
TESTED=vie/voe_auto_test, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/1058004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2013-01-22 12:31:01 +00:00
parent 8382ad557b
commit 5accd370e7
10 changed files with 150 additions and 120 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
View File

@@ -17,7 +17,7 @@ namespace webrtc {
class MockRTPReceiverVideo : public RTPReceiverVideo { class MockRTPReceiverVideo : public RTPReceiverVideo {
public: public:
MockRTPReceiverVideo() : RTPReceiverVideo(0, NULL, NULL, NULL) {} MockRTPReceiverVideo() : RTPReceiverVideo(0, NULL, NULL) {}
MOCK_METHOD1(ChangeUniqueId, MOCK_METHOD1(ChangeUniqueId,
void(const WebRtc_Word32 id)); void(const WebRtc_Word32 id));
MOCK_METHOD3(ReceiveRecoveredPacketCallback, MOCK_METHOD3(ReceiveRecoveredPacketCallback,

58
webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
View File

@@ -32,22 +32,16 @@ using ModuleRTPUtility::StringCompare;
using ModuleRTPUtility::VideoPayload; using ModuleRTPUtility::VideoPayload;
RTPReceiver::RTPReceiver(const WebRtc_Word32 id, RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
const bool audio,
Clock* clock, Clock* clock,
ModuleRtpRtcpImpl* owner, ModuleRtpRtcpImpl* owner,
RtpAudioFeedback* incoming_audio_messages_callback, RtpAudioFeedback* incoming_audio_messages_callback,
RtpData* incoming_payload_callback, RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback) RtpFeedback* incoming_messages_callback,
RTPReceiverStrategy* rtp_media_receiver,
RTPPayloadRegistry* rtp_payload_registry)
: Bitrate(clock), : Bitrate(clock),
rtp_payload_registry_(id), rtp_payload_registry_(rtp_payload_registry),
// TODO(phoglund): Remove hacks requiring direct access to the rtp_media_receiver_(rtp_media_receiver),
// audio receiver and only instantiate one of these directly into the
// rtp_media_receiver_ field. Right now an audio receiver carries around a
// video handler and vice versa, which doesn't make sense.
rtp_receiver_audio_(new RTPReceiverAudio(
id, incoming_payload_callback, incoming_audio_messages_callback)),
rtp_receiver_video_(new RTPReceiverVideo(
id, &rtp_payload_registry_, incoming_payload_callback, owner)),
id_(id), id_(id),
rtp_rtcp_(*owner), rtp_rtcp_(*owner),
cb_rtp_feedback_(incoming_messages_callback), cb_rtp_feedback_(incoming_messages_callback),
@@ -103,15 +97,6 @@ RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
incoming_messages_callback && incoming_messages_callback &&
incoming_payload_callback); incoming_payload_callback);
if (audio) {
rtp_media_receiver_ = rtp_receiver_audio_.get();
} else {
rtp_media_receiver_ = rtp_receiver_video_.get();
}
// TODO(phoglund): Get rid of this silly circular dependency between the
// payload manager and the video RTP receiver.
rtp_payload_registry_.set_rtp_media_receiver(rtp_media_receiver_);
memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_)); memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
memset(current_remote_energy_, 0, sizeof(current_remote_energy_)); memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
@@ -140,11 +125,11 @@ WebRtc_UWord32 RTPReceiver::MaxConfiguredBitrate() const {
} }
bool RTPReceiver::REDPayloadType(const WebRtc_Word8 payload_type) const { bool RTPReceiver::REDPayloadType(const WebRtc_Word8 payload_type) const {
return rtp_payload_registry_.red_payload_type() == payload_type; return rtp_payload_registry_->red_payload_type() == payload_type;
} }
WebRtc_Word8 RTPReceiver::REDPayloadType() const { WebRtc_Word8 RTPReceiver::REDPayloadType() const {
return rtp_payload_registry_.red_payload_type(); return rtp_payload_registry_->red_payload_type();
} }
WebRtc_Word32 RTPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeout_ms) { WebRtc_Word32 RTPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeout_ms) {
@@ -176,7 +161,7 @@ void RTPReceiver::PacketTimeout() {
if (now - last_receive_time_ > packet_timeout_ms_) { if (now - last_receive_time_ > packet_timeout_ms_) {
packet_time_out = true; packet_time_out = true;
last_receive_time_ = 0; // Only one callback. last_receive_time_ = 0; // Only one callback.
rtp_payload_registry_.ResetLastReceivedPayloadTypes(); rtp_payload_registry_->ResetLastReceivedPayloadTypes();
} }
} }
if (packet_time_out) { if (packet_time_out) {
@@ -226,14 +211,14 @@ WebRtc_Word32 RTPReceiver::RegisterReceivePayload(
const WebRtc_UWord8 channels, const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) { const WebRtc_UWord32 rate) {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_.RegisterReceivePayload( return rtp_payload_registry_->RegisterReceivePayload(
payload_name, payload_type, frequency, channels, rate); payload_name, payload_type, frequency, channels, rate);
} }
WebRtc_Word32 RTPReceiver::DeRegisterReceivePayload( WebRtc_Word32 RTPReceiver::DeRegisterReceivePayload(
const WebRtc_Word8 payload_type) { const WebRtc_Word8 payload_type) {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_.DeRegisterReceivePayload(payload_type); return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
} }
WebRtc_Word32 RTPReceiver::ReceivePayloadType( WebRtc_Word32 RTPReceiver::ReceivePayloadType(
@@ -243,7 +228,7 @@ WebRtc_Word32 RTPReceiver::ReceivePayloadType(
const WebRtc_UWord32 rate, const WebRtc_UWord32 rate,
WebRtc_Word8* payload_type) const { WebRtc_Word8* payload_type) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_.ReceivePayloadType( return rtp_payload_registry_->ReceivePayloadType(
payload_name, frequency, channels, rate, payload_type); payload_name, frequency, channels, rate, payload_type);
} }
@@ -254,7 +239,7 @@ WebRtc_Word32 RTPReceiver::ReceivePayload(
WebRtc_UWord8* channels, WebRtc_UWord8* channels,
WebRtc_UWord32* rate) const { WebRtc_UWord32* rate) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_.ReceivePayload( return rtp_payload_registry_->ReceivePayload(
payload_type, payload_name, frequency, channels, rate); payload_type, payload_name, frequency, channels, rate);
} }
@@ -627,7 +612,7 @@ WebRtc_UWord32 RTPReceiver::PayloadTypeToPayload(
const WebRtc_UWord8 payload_type, const WebRtc_UWord8 payload_type,
Payload*& payload) const { Payload*& payload) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
return rtp_payload_registry_.PayloadTypeToPayload(payload_type, payload); return rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload);
} }
// Compute time stamp of the last incoming packet that is the first packet of // Compute time stamp of the last incoming packet that is the first packet of
@@ -689,7 +674,7 @@ void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtp_header) {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
WebRtc_Word8 last_received_payload_type = WebRtc_Word8 last_received_payload_type =
rtp_payload_registry_.last_received_payload_type(); rtp_payload_registry_->last_received_payload_type();
if (ssrc_ != rtp_header->header.ssrc || if (ssrc_ != rtp_header->header.ssrc ||
(last_received_payload_type == -1 && ssrc_ == 0)) { (last_received_payload_type == -1 && ssrc_ == 0)) {
// We need the payload_type_ to make the call if the remote SSRC is 0. // We need the payload_type_ to make the call if the remote SSRC is 0.
@@ -709,7 +694,7 @@ void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtp_header) {
re_initialize_decoder = true; re_initialize_decoder = true;
Payload* payload; Payload* payload;
if (rtp_payload_registry_.PayloadTypeToPayload( if (rtp_payload_registry_->PayloadTypeToPayload(
rtp_header->header.payloadType, payload) != 0) { rtp_header->header.payloadType, payload) != 0) {
return; return;
} }
@@ -767,7 +752,7 @@ WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
WebRtc_Word8 last_received_payload_type = WebRtc_Word8 last_received_payload_type =
rtp_payload_registry_.last_received_payload_type(); rtp_payload_registry_->last_received_payload_type();
if (payload_type != last_received_payload_type) { if (payload_type != last_received_payload_type) {
if (REDPayloadType(payload_type)) { if (REDPayloadType(payload_type)) {
// Get the real codec payload type. // Get the real codec payload type.
@@ -803,7 +788,7 @@ WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
} }
Payload* payload; Payload* payload;
if (rtp_payload_registry_.PayloadTypeToPayload(payload_type, if (rtp_payload_registry_->PayloadTypeToPayload(payload_type,
payload) != 0) { payload) != 0) {
// Not a registered payload type. // Not a registered payload type.
return -1; return -1;
@@ -812,7 +797,7 @@ WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
rtp_payload_registry_.set_last_received_payload_type(payload_type); rtp_payload_registry_->set_last_received_payload_type(payload_type);
re_initialize_decoder = true; re_initialize_decoder = true;
@@ -825,7 +810,7 @@ WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
re_initialize_decoder = false; re_initialize_decoder = false;
} else { } else {
bool media_type_unchanged = bool media_type_unchanged =
rtp_payload_registry_.ReportMediaPayloadType(payload_type); rtp_payload_registry_->ReportMediaPayloadType(payload_type);
if (media_type_unchanged) { if (media_type_unchanged) {
// Only reset the decoder if the media codec type has changed. // Only reset the decoder if the media codec type has changed.
re_initialize_decoder = false; re_initialize_decoder = false;
@@ -860,9 +845,8 @@ void RTPReceiver::CheckCSRC(const WebRtcRTPHeader* rtp_header) {
{ {
CriticalSectionScoped lock(critical_section_rtp_receiver_); CriticalSectionScoped lock(critical_section_rtp_receiver_);
if (rtp_receiver_audio_->TelephoneEventPayloadType( if (!rtp_media_receiver_->ShouldReportCsrcChanges(
rtp_header->header.payloadType)) { rtp_header->header.payloadType)) {
// Don't do this for DTMF packets.
return; return;
} }
num_energy_ = rtp_header->type.Audio.numEnergy; num_energy_ = rtp_header->type.Audio.numEnergy;

20
webrtc/modules/rtp_rtcp/source/rtp_receiver.h
View File

@@ -34,15 +34,17 @@ class RTPReceiverStrategy;
class RTPReceiver : public Bitrate { class RTPReceiver : public Bitrate {
public: public:
// Callbacks passed in here may not be NULL (use Null object callbacks if you // Callbacks passed in here may not be NULL (use Null Object callbacks if you
// want callbacks to do nothing). // want callbacks to do nothing). This class takes ownership of the media
// receiver but nothing else.
RTPReceiver(const WebRtc_Word32 id, RTPReceiver(const WebRtc_Word32 id,
const bool audio,
Clock* clock, Clock* clock,
ModuleRtpRtcpImpl* owner, ModuleRtpRtcpImpl* owner,
RtpAudioFeedback* incoming_audio_messages_callback, RtpAudioFeedback* incoming_audio_messages_callback,
RtpData* incoming_payload_callback, RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback); RtpFeedback* incoming_messages_callback,
RTPReceiverStrategy* rtp_media_receiver,
RTPPayloadRegistry* rtp_payload_registry);
virtual ~RTPReceiver(); virtual ~RTPReceiver();
@@ -154,10 +156,6 @@ class RTPReceiver : public Bitrate {
void RTXStatus(bool* enable, WebRtc_UWord32* ssrc) const; void RTXStatus(bool* enable, WebRtc_UWord32* ssrc) const;
RTPReceiverAudio* GetAudioReceiver() const {
return rtp_receiver_audio_.get();
}
virtual WebRtc_Word8 REDPayloadType() const; virtual WebRtc_Word8 REDPayloadType() const;
bool HaveNotReceivedPackets() const; bool HaveNotReceivedPackets() const;
@@ -187,10 +185,8 @@ class RTPReceiver : public Bitrate {
bool ProcessNACKBitRate(WebRtc_UWord32 now); bool ProcessNACKBitRate(WebRtc_UWord32 now);
private: private:
RTPPayloadRegistry rtp_payload_registry_; RTPPayloadRegistry* rtp_payload_registry_;
scoped_ptr<RTPReceiverAudio> rtp_receiver_audio_; scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
scoped_ptr<RTPReceiverVideo> rtp_receiver_video_;
RTPReceiverStrategy* rtp_media_receiver_;
WebRtc_Word32 id_; WebRtc_Word32 id_;
ModuleRtpRtcpImpl& rtp_rtcp_; ModuleRtpRtcpImpl& rtp_rtcp_;

7
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
View File

@@ -148,6 +148,12 @@ RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType,
return false; return false;
} }
bool RTPReceiverAudio::ShouldReportCsrcChanges(
WebRtc_UWord8 payload_type) const {
// Don't do this for DTMF packets, otherwise it's fine.
return !TelephoneEventPayloadType(payload_type);
}
/* /*
Sample based or frame based codecs based on RFC 3551 Sample based or frame based codecs based on RFC 3551
@@ -182,7 +188,6 @@ RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType,
G7221 frame N/A G7221 frame N/A
*/ */
ModuleRTPUtility::Payload* RTPReceiverAudio::CreatePayloadType( ModuleRTPUtility::Payload* RTPReceiverAudio::CreatePayloadType(
const char payloadName[RTP_PAYLOAD_NAME_SIZE], const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType, const WebRtc_Word8 payloadType,

2
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
View File

@@ -76,6 +76,8 @@ public:
ModuleRTPUtility::Payload* payload, ModuleRTPUtility::Payload* payload,
const WebRtc_UWord32 rate) const; const WebRtc_UWord32 rate) const;
bool ShouldReportCsrcChanges(WebRtc_UWord8 payload_type) const;
ModuleRTPUtility::Payload* CreatePayloadType( ModuleRTPUtility::Payload* CreatePayloadType(
const char payloadName[RTP_PAYLOAD_NAME_SIZE], const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType, const WebRtc_Word8 payloadType,

4
webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
View File

@@ -68,6 +68,10 @@ class RTPReceiverStrategy {
ModuleRTPUtility::Payload* payload, ModuleRTPUtility::Payload* payload,
const WebRtc_UWord32 rate) const = 0; const WebRtc_UWord32 rate) const = 0;
// Returns true if we should report CSRC changes for this payload type.
// TODO(phoglund): should move out of here along with other payload stuff.
virtual bool ShouldReportCsrcChanges(WebRtc_UWord8 payload_type) const = 0;
// Creates a media-specific payload instance from the provided parameters. // Creates a media-specific payload instance from the provided parameters.
virtual ModuleRTPUtility::Payload* CreatePayloadType( virtual ModuleRTPUtility::Payload* CreatePayloadType(
const char payload_name[RTP_PAYLOAD_NAME_SIZE], const char payload_name[RTP_PAYLOAD_NAME_SIZE],

9
webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
View File

@@ -30,8 +30,7 @@ WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
RTPReceiverVideo::RTPReceiverVideo( RTPReceiverVideo::RTPReceiverVideo(
const WebRtc_Word32 id, const WebRtc_Word32 id,
const RTPPayloadRegistry* rtpRtpPayloadRegistry, const RTPPayloadRegistry* rtpRtpPayloadRegistry,
RtpData* data_callback, RtpData* data_callback)
ModuleRtpRtcpImpl* owner)
: RTPReceiverStrategy(data_callback), : RTPReceiverStrategy(data_callback),
_id(id), _id(id),
_rtpRtpPayloadRegistry(rtpRtpPayloadRegistry), _rtpRtpPayloadRegistry(rtpRtpPayloadRegistry),
@@ -46,6 +45,12 @@ RTPReceiverVideo::~RTPReceiverVideo() {
delete _receiveFEC; delete _receiveFEC;
} }
bool RTPReceiverVideo::ShouldReportCsrcChanges(
WebRtc_UWord8 payload_type) const {
// Always do this for video packets.
return true;
}
ModuleRTPUtility::Payload* RTPReceiverVideo::CreatePayloadType( ModuleRTPUtility::Payload* RTPReceiverVideo::CreatePayloadType(
const char payloadName[RTP_PAYLOAD_NAME_SIZE], const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType, const WebRtc_Word8 payloadType,

5
webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
View File

@@ -29,8 +29,7 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
public: public:
RTPReceiverVideo(const WebRtc_Word32 id, RTPReceiverVideo(const WebRtc_Word32 id,
const RTPPayloadRegistry* rtp_payload_registry, const RTPPayloadRegistry* rtp_payload_registry,
RtpData* data_callback, RtpData* data_callback);
ModuleRtpRtcpImpl* owner);
virtual ~RTPReceiverVideo(); virtual ~RTPReceiverVideo();
@@ -57,6 +56,8 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
ModuleRTPUtility::Payload* payload, ModuleRTPUtility::Payload* payload,
const WebRtc_UWord32 rate) const; const WebRtc_UWord32 rate) const;
bool ShouldReportCsrcChanges(WebRtc_UWord8 payload_type) const;
ModuleRTPUtility::Payload* CreatePayloadType( ModuleRTPUtility::Payload* CreatePayloadType(
const char payloadName[RTP_PAYLOAD_NAME_SIZE], const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType, const WebRtc_Word8 payloadType,

156
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@@ -88,20 +88,18 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
} }
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtp_sender_(configuration.id, : rtp_payload_registry_(configuration.id),
rtp_sender_(configuration.id,
configuration.audio, configuration.audio,
configuration.clock, configuration.clock,
configuration.outgoing_transport, configuration.outgoing_transport,
configuration.audio_messages, configuration.audio_messages,
configuration.paced_sender), configuration.paced_sender),
rtp_receiver_(configuration.id, configuration.audio, configuration.clock,
this, configuration.audio_messages,
configuration.incoming_data,
configuration.incoming_messages),
rtcp_sender_(configuration.id, configuration.audio, configuration.clock, rtcp_sender_(configuration.id, configuration.audio, configuration.clock,
this), this),
rtcp_receiver_(configuration.id, configuration.clock, this), rtcp_receiver_(configuration.id, configuration.clock, this),
clock_(configuration.clock), clock_(configuration.clock),
rtp_telephone_event_handler_(NULL),
id_(configuration.id), id_(configuration.id),
audio_(configuration.audio), audio_(configuration.audio),
collision_detected_(false), collision_detected_(false),
@@ -111,11 +109,11 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.clock->TimeInMilliseconds()), configuration.clock->TimeInMilliseconds()),
packet_overhead_(28), // IPV4 UDP. packet_overhead_(28), // IPV4 UDP.
critical_section_module_ptrs_( critical_section_module_ptrs_(
CriticalSectionWrapper::CreateCriticalSection()), CriticalSectionWrapper::CreateCriticalSection()),
critical_section_module_ptrs_feedback_( critical_section_module_ptrs_feedback_(
CriticalSectionWrapper::CreateCriticalSection()), CriticalSectionWrapper::CreateCriticalSection()),
default_module_( default_module_(
static_cast<ModuleRtpRtcpImpl*>(configuration.default_module)), static_cast<ModuleRtpRtcpImpl*>(configuration.default_module)),
dead_or_alive_active_(false), dead_or_alive_active_(false),
dead_or_alive_timeout_ms_(0), dead_or_alive_timeout_ms_(0),
dead_or_alive_last_timer_(0), dead_or_alive_last_timer_(0),
@@ -130,6 +128,30 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
, plot1_(NULL) , plot1_(NULL)
#endif #endif
{ {
RTPReceiverStrategy* rtp_receiver_strategy;
if (configuration.audio) {
// If audio, we need to be able to handle telephone events too, so stash
// away the audio receiver for those situations.
rtp_telephone_event_handler_ =
new RTPReceiverAudio(configuration.id, configuration.incoming_data,
configuration.audio_messages);
rtp_receiver_strategy = rtp_telephone_event_handler_;
} else {
rtp_receiver_strategy =
new RTPReceiverVideo(configuration.id, &rtp_payload_registry_,
configuration.incoming_data);
}
// TODO(phoglund): Get rid of this silly circular dependency between the
// payload manager and the video RTP receiver.
rtp_payload_registry_.set_rtp_media_receiver(rtp_receiver_strategy);
rtp_receiver_.reset(new RTPReceiver(
configuration.id, configuration.clock, this,
configuration.audio_messages, configuration.incoming_data,
configuration.incoming_messages, rtp_receiver_strategy,
&rtp_payload_registry_));
send_video_codec_.codecType = kVideoCodecUnknown; send_video_codec_.codecType = kVideoCodecUnknown;
if (default_module_) { if (default_module_) {
@@ -223,14 +245,14 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
if (now >= if (now >=
last_packet_timeout_process_time_ + kRtpRtcpPacketTimeoutProcessTimeMs) { last_packet_timeout_process_time_ + kRtpRtcpPacketTimeoutProcessTimeMs) {
rtp_receiver_.PacketTimeout(); rtp_receiver_->PacketTimeout();
rtcp_receiver_.PacketTimeout(); rtcp_receiver_.PacketTimeout();
last_packet_timeout_process_time_ = now; last_packet_timeout_process_time_ = now;
} }
if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) { if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) {
rtp_sender_.ProcessBitrate(); rtp_sender_.ProcessBitrate();
rtp_receiver_.ProcessBitrate(); rtp_receiver_->ProcessBitrate();
last_bitrate_process_time_ = now; last_bitrate_process_time_ = now;
} }
@@ -276,7 +298,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
if (remote_bitrate_) { if (remote_bitrate_) {
// TODO(mflodman) Remove this and let this be propagated by CallStats. // TODO(mflodman) Remove this and let this be propagated by CallStats.
remote_bitrate_->SetRtt(max_rtt); remote_bitrate_->SetRtt(max_rtt);
remote_bitrate_->UpdateEstimate(rtp_receiver_.SSRC(), now); remote_bitrate_->UpdateEstimate(rtp_receiver_->SSRC(), now);
if (TMMBR()) { if (TMMBR()) {
unsigned int target_bitrate = 0; unsigned int target_bitrate = 0;
std::vector<unsigned int> ssrcs; std::vector<unsigned int> ssrcs;
@@ -309,7 +331,7 @@ void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
if (rtcp_receiver_.LastReceived() + 12000 > now) { if (rtcp_receiver_.LastReceived() + 12000 > now) {
RTCPalive = true; RTCPalive = true;
} }
rtp_receiver_.ProcessDeadOrAlive(RTCPalive, now); rtp_receiver_->ProcessDeadOrAlive(RTCPalive, now);
} }
} }
} }
@@ -363,7 +385,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetPacketTimeout(
rtp_timeout_ms, rtp_timeout_ms,
rtcp_timeout_ms); rtcp_timeout_ms);
if (rtp_receiver_.SetPacketTimeout(rtp_timeout_ms) == 0) { if (rtp_receiver_->SetPacketTimeout(rtp_timeout_ms) == 0) {
return rtcp_receiver_.SetPacketTimeout(rtcp_timeout_ms); return rtcp_receiver_.SetPacketTimeout(rtcp_timeout_ms);
} }
return -1; return -1;
@@ -376,7 +398,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload(
id_, id_,
"RegisterReceivePayload(voice_codec)"); "RegisterReceivePayload(voice_codec)");
return rtp_receiver_.RegisterReceivePayload( return rtp_receiver_->RegisterReceivePayload(
voice_codec.plname, voice_codec.plname,
voice_codec.pltype, voice_codec.pltype,
voice_codec.plfreq, voice_codec.plfreq,
@@ -391,11 +413,11 @@ WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload(
id_, id_,
"RegisterReceivePayload(video_codec)"); "RegisterReceivePayload(video_codec)");
return rtp_receiver_.RegisterReceivePayload(video_codec.plName, return rtp_receiver_->RegisterReceivePayload(video_codec.plName,
video_codec.plType, video_codec.plType,
90000, 90000,
0, 0,
video_codec.maxBitrate); video_codec.maxBitrate);
} }
WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType( WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
@@ -406,7 +428,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
id_, id_,
"ReceivePayloadType(voice_codec)"); "ReceivePayloadType(voice_codec)");
return rtp_receiver_.ReceivePayloadType( return rtp_receiver_->ReceivePayloadType(
voice_codec.plname, voice_codec.plname,
voice_codec.plfreq, voice_codec.plfreq,
voice_codec.channels, voice_codec.channels,
@@ -422,11 +444,11 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
id_, id_,
"ReceivePayloadType(video_codec)"); "ReceivePayloadType(video_codec)");
return rtp_receiver_.ReceivePayloadType(video_codec.plName, return rtp_receiver_->ReceivePayloadType(video_codec.plName,
90000, 90000,
0, 0,
video_codec.maxBitrate, video_codec.maxBitrate,
pl_type); pl_type);
} }
WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterReceivePayload( WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterReceivePayload(
@@ -437,7 +459,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterReceivePayload(
"DeRegisterReceivePayload(%d)", "DeRegisterReceivePayload(%d)",
payload_type); payload_type);
return rtp_receiver_.DeRegisterReceivePayload(payload_type); return rtp_receiver_->DeRegisterReceivePayload(payload_type);
} }
// Get the currently configured SSRC filter. // Get the currently configured SSRC filter.
@@ -445,7 +467,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SSRCFilter(
WebRtc_UWord32& allowed_ssrc) const { WebRtc_UWord32& allowed_ssrc) const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SSRCFilter()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SSRCFilter()");
return rtp_receiver_.SSRCFilter(allowed_ssrc); return rtp_receiver_->SSRCFilter(allowed_ssrc);
} }
// Set a SSRC to be used as a filter for incoming RTP streams. // Set a SSRC to be used as a filter for incoming RTP streams.
@@ -465,21 +487,21 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRCFilter(
"SetSSRCFilter(disable)"); "SetSSRCFilter(disable)");
} }
return rtp_receiver_.SetSSRCFilter(enable, allowed_ssrc); return rtp_receiver_->SetSSRCFilter(enable, allowed_ssrc);
} }
// Get last received remote timestamp. // Get last received remote timestamp.
WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteTimestamp() const { WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteTimestamp() const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteTimestamp()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteTimestamp()");
return rtp_receiver_.TimeStamp(); return rtp_receiver_->TimeStamp();
} }
int64_t ModuleRtpRtcpImpl::LocalTimeOfRemoteTimeStamp() const { int64_t ModuleRtpRtcpImpl::LocalTimeOfRemoteTimeStamp() const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
"LocalTimeOfRemoteTimeStamp()"); "LocalTimeOfRemoteTimeStamp()");
return rtp_receiver_.LastReceivedTimeMs(); return rtp_receiver_->LastReceivedTimeMs();
} }
// Get the current estimated remote timestamp. // Get the current estimated remote timestamp.
@@ -490,14 +512,14 @@ WebRtc_Word32 ModuleRtpRtcpImpl::EstimatedRemoteTimeStamp(
id_, id_,
"EstimatedRemoteTimeStamp()"); "EstimatedRemoteTimeStamp()");
return rtp_receiver_.EstimatedRemoteTimeStamp(timestamp); return rtp_receiver_->EstimatedRemoteTimeStamp(timestamp);
} }
// Get incoming SSRC. // Get incoming SSRC.
WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteSSRC() const { WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteSSRC() const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteSSRC()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteSSRC()");
return rtp_receiver_.SSRC(); return rtp_receiver_->SSRC();
} }
// Get remote CSRC // Get remote CSRC
@@ -505,7 +527,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCSRCs(
WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const { WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteCSRCs()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteCSRCs()");
return rtp_receiver_.CSRCs(arr_of_csrc); return rtp_receiver_->CSRCs(arr_of_csrc);
} }
WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXSendStatus( WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXSendStatus(
@@ -525,13 +547,13 @@ WebRtc_Word32 ModuleRtpRtcpImpl::RTXSendStatus(bool* enable,
WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXReceiveStatus( WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXReceiveStatus(
const bool enable, const bool enable,
const WebRtc_UWord32 ssrc) { const WebRtc_UWord32 ssrc) {
rtp_receiver_.SetRTXStatus(enable, ssrc); rtp_receiver_->SetRTXStatus(enable, ssrc);
return 0; return 0;
} }
WebRtc_Word32 ModuleRtpRtcpImpl::RTXReceiveStatus(bool* enable, WebRtc_Word32 ModuleRtpRtcpImpl::RTXReceiveStatus(bool* enable,
WebRtc_UWord32* ssrc) const { WebRtc_UWord32* ssrc) const {
rtp_receiver_.RTXStatus(enable, ssrc); rtp_receiver_->RTXStatus(enable, ssrc);
return 0; return 0;
} }
@@ -593,7 +615,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::IncomingPacket(
memset(&rtp_header, 0, sizeof(rtp_header)); memset(&rtp_header, 0, sizeof(rtp_header));
RtpHeaderExtensionMap map; RtpHeaderExtensionMap map;
rtp_receiver_.GetHeaderExtensionMapCopy(&map); rtp_receiver_->GetHeaderExtensionMapCopy(&map);
const bool valid_rtpheader = rtp_parser.Parse(rtp_header, &map); const bool valid_rtpheader = rtp_parser.Parse(rtp_header, &map);
if (!valid_rtpheader) { if (!valid_rtpheader) {
@@ -603,9 +625,9 @@ WebRtc_Word32 ModuleRtpRtcpImpl::IncomingPacket(
"IncomingPacket invalid RTP header"); "IncomingPacket invalid RTP header");
return -1; return -1;
} }
return rtp_receiver_.IncomingRTPPacket(&rtp_header, return rtp_receiver_->IncomingRTPPacket(&rtp_header,
incoming_packet, incoming_packet,
incoming_packet_length); incoming_packet_length);
} }
} }
@@ -1147,7 +1169,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCNAME(
WebRtc_UWord16 ModuleRtpRtcpImpl::RemoteSequenceNumber() const { WebRtc_UWord16 ModuleRtpRtcpImpl::RemoteSequenceNumber() const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteSequenceNumber()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteSequenceNumber()");
return rtp_receiver_.SequenceNumber(); return rtp_receiver_->SequenceNumber();
} }
WebRtc_Word32 ModuleRtpRtcpImpl::RemoteNTP( WebRtc_Word32 ModuleRtpRtcpImpl::RemoteNTP(
@@ -1193,7 +1215,7 @@ void ModuleRtpRtcpImpl:: SetRtt(uint32_t rtt) {
WebRtc_Word32 ModuleRtpRtcpImpl::ResetStatisticsRTP() { WebRtc_Word32 ModuleRtpRtcpImpl::ResetStatisticsRTP() {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetStatisticsRTP()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetStatisticsRTP()");
return rtp_receiver_.ResetStatistics(); return rtp_receiver_->ResetStatistics();
} }
// Reset RTP data counters for the receiving side. // Reset RTP data counters for the receiving side.
@@ -1201,7 +1223,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ResetReceiveDataCountersRTP() {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
"ResetReceiveDataCountersRTP()"); "ResetReceiveDataCountersRTP()");
return rtp_receiver_.ResetDataCounters(); return rtp_receiver_->ResetDataCounters();
} }
// Reset RTP data counters for the sending side. // Reset RTP data counters for the sending side.
@@ -1253,7 +1275,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::StatisticsRTP(
WebRtc_UWord32 jitter_transmission_time_offset = 0; WebRtc_UWord32 jitter_transmission_time_offset = 0;
WebRtc_Word32 ret_val = rtp_receiver_.Statistics( WebRtc_Word32 ret_val = rtp_receiver_->Statistics(
fraction_lost, cum_lost, ext_max, jitter, max_jitter, fraction_lost, cum_lost, ext_max, jitter, max_jitter,
&jitter_transmission_time_offset, (rtcp_sender_.Status() == kRtcpOff)); &jitter_transmission_time_offset, (rtcp_sender_.Status() == kRtcpOff));
if (ret_val == -1) { if (ret_val == -1) {
@@ -1276,7 +1298,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::DataCountersRTP(
if (packets_sent) { if (packets_sent) {
*packets_sent = rtp_sender_.Packets(); *packets_sent = rtp_sender_.Packets();
} }
return rtp_receiver_.DataCounters(bytes_received, packets_received); return rtp_receiver_->DataCounters(bytes_received, packets_received);
} }
WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics( WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics(
@@ -1287,14 +1309,14 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics(
WebRtc_UWord32* jitter_transmission_time_offset) { WebRtc_UWord32* jitter_transmission_time_offset) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ReportBlockStatistics()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ReportBlockStatistics()");
WebRtc_Word32 missing = 0; WebRtc_Word32 missing = 0;
WebRtc_Word32 ret = rtp_receiver_.Statistics(fraction_lost, WebRtc_Word32 ret = rtp_receiver_->Statistics(fraction_lost,
cum_lost, cum_lost,
ext_max, ext_max,
jitter, jitter,
NULL, NULL,
jitter_transmission_time_offset, jitter_transmission_time_offset,
&missing, &missing,
true); true);
#ifdef MATLAB #ifdef MATLAB
if (plot1_ == NULL) { if (plot1_ == NULL) {
@@ -1397,12 +1419,12 @@ WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceiveRtpHeaderExtension( WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceiveRtpHeaderExtension(
const RTPExtensionType type, const RTPExtensionType type,
const WebRtc_UWord8 id) { const WebRtc_UWord8 id) {
return rtp_receiver_.RegisterRtpHeaderExtension(type, id); return rtp_receiver_->RegisterRtpHeaderExtension(type, id);
} }
WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterReceiveRtpHeaderExtension( WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterReceiveRtpHeaderExtension(
const RTPExtensionType type) { const RTPExtensionType type) {
return rtp_receiver_.DeregisterRtpHeaderExtension(type); return rtp_receiver_->DeregisterRtpHeaderExtension(type);
} }
// (TMMBR) Temporary Max Media Bit Rate. // (TMMBR) Temporary Max Media Bit Rate.
@@ -1472,7 +1494,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetNACKStatus(NACKMethod method) {
"SetNACKStatus(%u)", method); "SetNACKStatus(%u)", method);
nack_method_ = method; nack_method_ = method;
rtp_receiver_.SetNACKStatus(method); rtp_receiver_->SetNACKStatus(method);
return 0; return 0;
} }
@@ -1509,7 +1531,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nack_list,
return -1; return -1;
} }
WebRtc_UWord16 avg_rtt = 0; WebRtc_UWord16 avg_rtt = 0;
rtcp_receiver_.RTT(rtp_receiver_.SSRC(), NULL, &avg_rtt, NULL, NULL); rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL);
WebRtc_Word64 wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5. WebRtc_Word64 wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5.
if (wait_time == 5) { if (wait_time == 5) {
@@ -1578,7 +1600,9 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetTelephoneEventStatus(
" detect_end_of_tone:%d)", enable, forward_to_decoder, " detect_end_of_tone:%d)", enable, forward_to_decoder,
detect_end_of_tone); detect_end_of_tone);
return rtp_receiver_.GetAudioReceiver()->SetTelephoneEventStatus( assert(audio_);
assert(rtp_telephone_event_handler_);
return rtp_telephone_event_handler_->SetTelephoneEventStatus(
enable, forward_to_decoder, detect_end_of_tone); enable, forward_to_decoder, detect_end_of_tone);
} }
@@ -1586,7 +1610,9 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetTelephoneEventStatus(
bool ModuleRtpRtcpImpl::TelephoneEvent() const { bool ModuleRtpRtcpImpl::TelephoneEvent() const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "TelephoneEvent()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "TelephoneEvent()");
return rtp_receiver_.GetAudioReceiver()->TelephoneEvent(); assert(audio_);
assert(rtp_telephone_event_handler_);
return rtp_telephone_event_handler_->TelephoneEvent();
} }
// Is forwarding of out-band telephone events turned on/off? // Is forwarding of out-band telephone events turned on/off?
@@ -1594,7 +1620,9 @@ bool ModuleRtpRtcpImpl::TelephoneEventForwardToDecoder() const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
"TelephoneEventForwardToDecoder()"); "TelephoneEventForwardToDecoder()");
return rtp_receiver_.GetAudioReceiver()->TelephoneEventForwardToDecoder(); assert(audio_);
assert(rtp_telephone_event_handler_);
return rtp_telephone_event_handler_->TelephoneEventForwardToDecoder();
} }
// Send a TelephoneEvent tone using RFC 2833 (4733). // Send a TelephoneEvent tone using RFC 2833 (4733).
@@ -1646,9 +1674,9 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus(
id); id);
if (enable) { if (enable) {
rtp_receiver_.RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, id); rtp_receiver_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, id);
} else { } else {
rtp_receiver_.DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel); rtp_receiver_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
} }
return rtp_sender_.SetAudioLevelIndicationStatus(enable, id); return rtp_sender_.SetAudioLevelIndicationStatus(enable, id);
} }
@@ -1695,7 +1723,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendREDPayloadType(
} }
RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const { RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const {
return rtp_receiver_.VideoCodecType(); return rtp_receiver_->VideoCodecType();
} }
RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const { RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const {
@@ -1910,7 +1938,7 @@ void ModuleRtpRtcpImpl::SetRemoteSSRC(const WebRtc_UWord32 ssrc) {
} }
WebRtc_UWord32 ModuleRtpRtcpImpl::BitrateReceivedNow() const { WebRtc_UWord32 ModuleRtpRtcpImpl::BitrateReceivedNow() const {
return rtp_receiver_.BitrateNow(); return rtp_receiver_->BitrateNow();
} }
void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* total_rate, void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* total_rate,
@@ -2012,7 +2040,7 @@ void ModuleRtpRtcpImpl::OnReceivedNACK(
return; return;
} }
WebRtc_UWord16 avg_rtt = 0; WebRtc_UWord16 avg_rtt = 0;
rtcp_receiver_.RTT(rtp_receiver_.SSRC(), NULL, &avg_rtt, NULL, NULL); rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL);
rtp_sender_.OnReceivedNACK(nack_sequence_numbers_length, rtp_sender_.OnReceivedNACK(nack_sequence_numbers_length,
nack_sequence_numbers, nack_sequence_numbers,
avg_rtt); avg_rtt);

7
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@@ -17,6 +17,7 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -471,8 +472,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Only for internal testing. // Only for internal testing.
WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime); WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime);
RTPPayloadRegistry rtp_payload_registry_;
RTPSender rtp_sender_; RTPSender rtp_sender_;
RTPReceiver rtp_receiver_; scoped_ptr<RTPReceiver> rtp_receiver_;
RTCPSender rtcp_sender_; RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_; RTCPReceiver rtcp_receiver_;
@@ -482,6 +485,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
private: private:
int64_t RtcpReportInterval(); int64_t RtcpReportInterval();
RTPReceiverAudio* rtp_telephone_event_handler_;
WebRtc_Word32 id_; WebRtc_Word32 id_;
const bool audio_; const bool audio_;
bool collision_detected_; bool collision_detected_;