Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(224)

Unified Diff: webrtc/video/video_receive_stream.cc

Issue 1671893002: Remove ViEChannel calls for VideoReceiveStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: remove extra include + forward declare ProcessThread Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/video/video_receive_stream.cc
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index e18003d4e73bf845fa584fd8c93e5efd2761cc1d..692b3991b35a96426c11ed4a5fb4e7d8ca4ff8e4 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -174,7 +174,9 @@ VideoReceiveStream::VideoReceiveStream(
congestion_controller_->pacer(),
congestion_controller_->packet_router(),
1,
- false) {
+ false),
+ vie_receiver_(vie_channel_.vie_receiver()),
+ rtp_rtcp_(vie_channel_.rtp_rtcp()) {
LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
RTC_CHECK(vie_channel_.Init() == 0);
@@ -183,27 +185,26 @@ VideoReceiveStream::VideoReceiveStream(
call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
// TODO(pbos): This is not fine grained enough...
- vie_channel_.SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false,
- -1, -1);
+ vie_channel_.SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, -1,
+ -1);
RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
<< "A stream should not be configured with RTCP disabled. This value is "
"reserved for internal usage.";
- vie_channel_.SetRTCPMode(config_.rtp.rtcp_mode);
+ rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
RTC_DCHECK(config_.rtp.remote_ssrc != 0);
// TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
RTC_DCHECK(config_.rtp.local_ssrc != 0);
RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
+ rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
- vie_channel_.SetSSRC(config_.rtp.local_ssrc, kViEStreamTypeNormal, 0);
// TODO(pbos): Support multiple RTX, per video payload.
- Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
- for (; it != config_.rtp.rtx.end(); ++it) {
- RTC_DCHECK(it->second.ssrc != 0);
- RTC_DCHECK(it->second.payload_type != 0);
+ for (const auto& kv : config_.rtp.rtx) {
+ RTC_DCHECK(kv.second.ssrc != 0);
+ RTC_DCHECK(kv.second.payload_type != 0);
- vie_channel_.SetRemoteSSRCType(kViEStreamTypeRtx, it->second.ssrc);
- vie_channel_.SetRtxReceivePayloadType(it->second.payload_type, it->first);
+ vie_receiver_->SetRtxSsrc(kv.second.ssrc);
+ vie_channel_.SetRtxReceivePayloadType(kv.second.payload_type, kv.first);
}
// TODO(holmer): When Chrome no longer depends on this being false by default,
// always use the mapping and remove this whole codepath.
@@ -211,7 +212,7 @@ VideoReceiveStream::VideoReceiveStream(
config_.rtp.use_rtx_payload_mapping_on_restore);
congestion_controller_->SetChannelRembStatus(false, config_.rtp.remb,
- vie_channel_.rtp_rtcp());
+ rtp_rtcp_);
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
@@ -220,14 +221,13 @@ VideoReceiveStream::VideoReceiveStream(
RTC_DCHECK_GE(id, 1);
RTC_DCHECK_LE(id, 14);
if (extension == RtpExtension::kTOffset) {
- RTC_CHECK_EQ(0, vie_channel_.SetReceiveTimestampOffsetStatus(true, id));
+ RTC_CHECK(vie_receiver_->SetReceiveTimestampOffsetStatus(true, id));
} else if (extension == RtpExtension::kAbsSendTime) {
- RTC_CHECK_EQ(0, vie_channel_.SetReceiveAbsoluteSendTimeStatus(true, id));
+ RTC_CHECK(vie_receiver_->SetReceiveAbsoluteSendTimeStatus(true, id));
} else if (extension == RtpExtension::kVideoRotation) {
- RTC_CHECK_EQ(0, vie_channel_.SetReceiveVideoRotationStatus(true, id));
+ RTC_CHECK(vie_receiver_->SetReceiveVideoRotationStatus(true, id));
} else if (extension == RtpExtension::kTransportSequenceNumber) {
- RTC_CHECK_EQ(0,
- vie_channel_.SetReceiveTransportSequenceNumber(true, id));
+ RTC_CHECK(vie_receiver_->SetReceiveTransportSequenceNumber(true, id));
} else {
RTC_NOTREACHED() << "Unsupported RTP extension.";
}
@@ -258,7 +258,7 @@ VideoReceiveStream::VideoReceiveStream(
}
if (config.rtp.rtcp_xr.receiver_reference_time_report)
- vie_channel_.SetRtcpXrRrtrStatus(true);
+ rtp_rtcp_->SetRtcpXrRrtrStatus(true);
vie_channel_.RegisterReceiveStatisticsProxy(&stats_proxy_);
vie_channel_.RegisterReceiveChannelRtcpStatisticsCallback(&stats_proxy_);
@@ -275,20 +275,18 @@ VideoReceiveStream::VideoReceiveStream(
<< "Duplicate payload type (" << decoder.payload_type
<< ") for different decoders.";
decoder_payload_types.insert(decoder.payload_type);
- vie_channel_.RegisterExternalDecoder(decoder.payload_type,
- decoder.decoder);
+ vcm_->RegisterExternalDecoder(decoder.decoder, decoder.payload_type);
VideoCodec codec = CreateDecoderVideoCodec(decoder);
RTC_CHECK_EQ(0, vie_channel_.SetReceiveCodec(codec));
}
+ vcm_->SetRenderDelay(config.render_delay_ms);
incoming_video_stream_.SetExpectedRenderDelay(config.render_delay_ms);
- vie_channel_.SetExpectedRenderDelay(config.render_delay_ms);
+ vcm_->RegisterPreDecodeImageCallback(this);
incoming_video_stream_.SetExternalCallback(this);
vie_channel_.SetIncomingVideoStream(&incoming_video_stream_);
-
- vie_channel_.RegisterPreDecodeImageCallback(this);
vie_channel_.RegisterPreRenderCallback(this);
process_thread_->RegisterModule(vcm_.get());
@@ -299,15 +297,13 @@ VideoReceiveStream::~VideoReceiveStream() {
incoming_video_stream_.Stop();
process_thread_->DeRegisterModule(vcm_.get());
vie_channel_.RegisterPreRenderCallback(nullptr);
- vie_channel_.RegisterPreDecodeImageCallback(nullptr);
+ vcm_->RegisterPreDecodeImageCallback(nullptr);
call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
- congestion_controller_->SetChannelRembStatus(false, false,
- vie_channel_.rtp_rtcp());
+ congestion_controller_->SetChannelRembStatus(false, false, rtp_rtcp_);
- uint32_t remote_ssrc = vie_channel_.GetRemoteSSRC();
congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config_))
- ->RemoveStream(remote_ssrc);
+ ->RemoveStream(vie_receiver_->GetRemoteSsrc());
}
void VideoReceiveStream::Start() {
@@ -338,13 +334,13 @@ VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
}
bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
- return vie_channel_.ReceivedRTCPPacket(packet, length) == 0;
+ return vie_receiver_->DeliverRtcp(packet, length);
}
bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
- return vie_channel_.ReceivedRTPPacket(packet, length, packet_time) == 0;
+ return vie_receiver_->DeliverRtp(packet, length, packet_time);
}
void VideoReceiveStream::FrameCallback(VideoFrame* video_frame) {
@@ -389,8 +385,8 @@ int32_t VideoReceiveStream::Encoded(
}
void VideoReceiveStream::SignalNetworkState(NetworkState state) {
- vie_channel_.SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
- : RtcpMode::kOff);
+ rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
+ : RtcpMode::kOff);
}
} // namespace internal

Powered by Google App Engine
This is Rietveld 408576698