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Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.

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The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2013-01-14 10:01:55 +00:00
parent 49273ffa79
commit a22a9bd9ca
12 changed files with 184 additions and 152 deletions
Download Patch File Download Diff File Expand all files Collapse all files

126
webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
View File

@ -35,13 +35,21 @@ RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
const bool audio,
RtpRtcpClock* clock,
ModuleRtpRtcpImpl* owner,
RtpAudioFeedback* incoming_messages_callback)
RtpAudioFeedback* incoming_audio_messages_callback,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback)
: Bitrate(clock),
// TODO(phoglund): Remove hacks requiring direct access to the
// 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, this, incoming_payload_callback, owner)),
id_(id),
rtp_rtcp_(*owner),
critical_section_cbs_(CriticalSectionWrapper::CreateCriticalSection()),
cb_rtp_feedback_(NULL),
cb_rtp_data_(NULL),
cb_rtp_feedback_(incoming_messages_callback),
critical_section_rtp_receiver_(
CriticalSectionWrapper::CreateCriticalSection()),
@ -94,14 +102,9 @@ RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
nack_method_(kNackOff),
rtx_(false),
ssrc_rtx_(0) {
// TODO(phoglund): Remove hacks requiring direct access to the 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, this,
incoming_messages_callback);
rtp_receiver_video_ = new RTPReceiverVideo(id, this, owner);
assert(incoming_audio_messages_callback &&
incoming_messages_callback &&
incoming_payload_callback);
if (audio) {
rtp_media_receiver_ = rtp_receiver_audio_;
} else {
@ -115,13 +118,10 @@ RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
}
RTPReceiver::~RTPReceiver() {
if (cb_rtp_feedback_) {
for (int i = 0; i < num_csrcs_; ++i) {
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
false);
}
for (int i = 0; i < num_csrcs_; ++i) {
cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
false);
}
delete critical_section_cbs_;
delete critical_section_rtp_receiver_;
while (!payload_type_map_.empty()) {
@ -187,18 +187,13 @@ void RTPReceiver::PacketTimeout() {
last_received_media_payload_type_ = -1;
}
}
CriticalSectionScoped lock(critical_section_cbs_);
if (packet_time_out && cb_rtp_feedback_) {
if (packet_time_out) {
cb_rtp_feedback_->OnPacketTimeout(id_);
}
}
void RTPReceiver::ProcessDeadOrAlive(const bool rtcp_alive,
const WebRtc_Word64 now) {
if (cb_rtp_feedback_ == NULL) {
// No callback.
return;
}
RTPAliveType alive = kRtpDead;
if (last_receive_time_ + 1000 > now) {
@ -214,10 +209,7 @@ void RTPReceiver::ProcessDeadOrAlive(const bool rtcp_alive,
}
}
CriticalSectionScoped lock(critical_section_cbs_);
if (cb_rtp_feedback_) {
cb_rtp_feedback_->OnPeriodicDeadOrAlive(id_, alive);
}
cb_rtp_feedback_->OnPeriodicDeadOrAlive(id_, alive);
}
WebRtc_UWord16 RTPReceiver::PacketOHReceived() const {
@ -235,20 +227,6 @@ WebRtc_UWord32 RTPReceiver::ByteCountReceived() const {
return received_byte_count_;
}
WebRtc_Word32 RTPReceiver::RegisterIncomingRTPCallback(
RtpFeedback* incoming_messages_callback) {
CriticalSectionScoped lock(critical_section_cbs_);
cb_rtp_feedback_ = incoming_messages_callback;
return 0;
}
WebRtc_Word32 RTPReceiver::RegisterIncomingDataCallback(
RtpData* incoming_data_callback) {
CriticalSectionScoped lock(critical_section_cbs_);
cb_rtp_data_ = incoming_data_callback;
return 0;
}
WebRtc_Word32 RTPReceiver::RegisterReceivePayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payload_type,
@ -610,14 +588,11 @@ WebRtc_Word32 RTPReceiver::IncomingRTPPacket(
}
if (last_receive_time_ == 0) {
// Trigger only once.
CriticalSectionScoped lock(critical_section_cbs_);
if (cb_rtp_feedback_) {
if (length - rtp_header->header.headerLength == 0) {
// Keep-alive packet.
cb_rtp_feedback_->OnReceivedPacket(id_, kPacketKeepAlive);
} else {
cb_rtp_feedback_->OnReceivedPacket(id_, kPacketRtp);
}
if (length - rtp_header->header.headerLength == 0) {
// Keep-alive packet.
cb_rtp_feedback_->OnReceivedPacket(id_, kPacketKeepAlive);
} else {
cb_rtp_feedback_->OnReceivedPacket(id_, kPacketRtp);
}
}
WebRtc_Word8 first_payload_byte = 0;
@ -686,19 +661,6 @@ WebRtc_Word32 RTPReceiver::IncomingRTPPacket(
return ret_val;
}
// Implementation note: must not hold critsect when called!
WebRtc_Word32 RTPReceiver::CallbackOfReceivedPayloadData(
const WebRtc_UWord8* payload_data,
const WebRtc_UWord16 payload_size,
const WebRtcRTPHeader* rtp_header) {
CriticalSectionScoped lock(critical_section_cbs_);
if (cb_rtp_data_) {
return cb_rtp_data_->OnReceivedPayloadData(payload_data, payload_size,
rtp_header);
}
return -1;
}
// Implementation note: we expect to have the critical_section_rtp_receiver_
// critsect when we call this.
void RTPReceiver::UpdateStatistics(const WebRtcRTPHeader* rtp_header,
@ -989,21 +951,16 @@ void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtp_header) {
// We need to get this to our RTCP sender and receiver.
// We need to do this outside critical section.
rtp_rtcp_.SetRemoteSSRC(rtp_header->header.ssrc);
cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header->header.ssrc);
}
CriticalSectionScoped lock(critical_section_cbs_);
if (cb_rtp_feedback_) {
if (new_ssrc) {
cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header->header.ssrc);
}
if (re_initialize_decoder) {
if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
id_, rtp_header->header.payloadType, payload_name, frequency,
channels, rate)) {
// New stream, same codec.
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
"Failed to create decoder for payload type:%d",
rtp_header->header.payloadType);
}
if (re_initialize_decoder) {
if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
id_, rtp_header->header.payloadType, payload_name, frequency,
channels, rate)) {
// New stream, same codec.
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
"Failed to create decoder for payload type:%d",
rtp_header->header.payloadType);
}
}
}
@ -1104,13 +1061,10 @@ WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
} // End critsect.
if (re_initialize_decoder) {
CriticalSectionScoped lock(critical_section_cbs_);
if (cb_rtp_feedback_) {
if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
cb_rtp_feedback_, id_, payload_type, payload_name,
*specific_payload)) {
return -1; // Wrong payload type.
}
if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
cb_rtp_feedback_, id_, payload_type, payload_name,
*specific_payload)) {
return -1; // Wrong payload type.
}
}
return 0;
@ -1159,10 +1113,6 @@ void RTPReceiver::CheckCSRC(const WebRtcRTPHeader* rtp_header) {
}
} // End critsect.
CriticalSectionScoped lock(critical_section_cbs_);
if (cb_rtp_feedback_ == NULL) {
return;
}
bool have_called_callback = false;
// Search for new CSRC in old array.
for (WebRtc_UWord8 i = 0; i < rtp_header->header.numCSRCs; ++i) {