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Extract send-side ViEReceiver calls.

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Removes ViEChannel hop for DeliverRtcp and GetRemoteSsrc calls.

BUG=webrtc:5494
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1687333002 .

Cr-Commit-Position: refs/heads/master@{#11576}
This commit is contained in:
Peter Boström 2016-02-11 15:30:14 +01:00
parent 68da769ab9
commit 1f7d77f60c
4 changed files with 5 additions and 26 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/video/video_send_stream.cc
View File

@ -168,6 +168,7 @@ VideoSendStream::VideoSendStream(
congestion_controller_->packet_router(),
config_.rtp.ssrcs.size(),
true),
vie_receiver_(vie_channel_.vie_receiver()),
input_(&vie_encoder_,
config_.local_renderer,
&stats_proxy_,
@ -285,9 +286,8 @@ VideoSendStream::~VideoSendStream() {
// done before deleting the channel.
encoder_feedback_.RemoveEncoder(&vie_encoder_);
uint32_t remote_ssrc = vie_channel_.GetRemoteSSRC();
congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
remote_ssrc);
vie_receiver_->GetRemoteSsrc());
}
VideoCaptureInput* VideoSendStream::Input() {
@ -465,7 +465,7 @@ bool VideoSendStream::ReconfigureVideoEncoder(
}
bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
return vie_channel_.ReceivedRTCPPacket(packet, length) == 0;
return vie_receiver_->DeliverRtcp(packet, length);
}
VideoSendStream::Stats VideoSendStream::GetStats() {

1
webrtc/video/video_send_stream.h
View File

@ -98,6 +98,7 @@ class VideoSendStream : public webrtc::VideoSendStream,
VideoCodingModule* const vcm_;
EncoderStateFeedback encoder_feedback_;
ViEChannel vie_channel_;
ViEReceiver* const vie_receiver_;
VideoCaptureInput input_;
};
} // namespace internal

15
webrtc/video/vie_channel.cc
View File

@ -610,13 +610,9 @@ int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) {
return 0;
}
uint32_t ViEChannel::GetRemoteSSRC() {
RTC_DCHECK(sender_);
return vie_receiver_.GetRemoteSsrc();
}
int ViEChannel::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
RTC_DCHECK(sender_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->SetRtxSendPayloadType(payload_type, associated_payload_type);
SetRtxSendStatus(true);
@ -837,15 +833,6 @@ void ViEChannel::StopReceive() {
StopDecodeThread();
}
int32_t ViEChannel::ReceivedRTCPPacket(const void* rtcp_packet,
size_t rtcp_packet_length) {
RTC_DCHECK(sender_);
return vie_receiver_.DeliverRtcp(
reinterpret_cast<const uint8_t*>(rtcp_packet), rtcp_packet_length)
? 0
: -1;
}
int32_t ViEChannel::SetMTU(uint16_t mtu) {
RTC_DCHECK(sender_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)

9
webrtc/video/vie_channel.h
View File

@ -106,9 +106,6 @@ class ViEChannel : public VCMFrameTypeCallback,
// Gets SSRC for outgoing stream number |idx|.
int32_t GetLocalSSRC(uint8_t idx, unsigned int* ssrc);
// Gets SSRC for the incoming stream.
uint32_t GetRemoteSSRC();
int SetRtxSendPayloadType(int payload_type, int associated_payload_type);
void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
@ -170,12 +167,6 @@ class ViEChannel : public VCMFrameTypeCallback,
void StartReceive();
void StopReceive();
int32_t ReceivedRTPPacket(const void* rtp_packet,
const size_t rtp_packet_length,
const PacketTime& packet_time);
int32_t ReceivedRTCPPacket(const void* rtcp_packet,
const size_t rtcp_packet_length);
// Sets the maximum transfer unit size for the network link, i.e. including
// IP, UDP and RTP headers.
int32_t SetMTU(uint16_t mtu);