Chromium Code Reviews| Index: media/cast/net/cast_transport_sender_impl.cc |
| diff --git a/media/cast/net/cast_transport_sender_impl.cc b/media/cast/net/cast_transport_sender_impl.cc |
| index 3c8ef0069a63cee677f2fb37a16ad6db958a92c0..0e8ad7a58425911916f36d3d103c29745a33bcc9 100644 |
| --- a/media/cast/net/cast_transport_sender_impl.cc |
| +++ b/media/cast/net/cast_transport_sender_impl.cc |
| @@ -4,9 +4,14 @@ |
| #include "media/cast/net/cast_transport_sender_impl.h" |
| +#include <algorithm> |
| +#include <string> |
| + |
| #include "base/single_thread_task_runner.h" |
| #include "base/values.h" |
| #include "media/cast/net/cast_transport_defines.h" |
| +#include "media/cast/net/rtcp/receiver_rtcp_session.h" |
| +#include "media/cast/net/rtcp/sender_rtcp_session.h" |
| #include "media/cast/net/udp_transport.h" |
| #include "net/base/net_errors.h" |
| #include "net/base/network_interfaces.h" |
| @@ -36,7 +41,7 @@ int LookupOptionWithDefault(const base::DictionaryValue& options, |
| } else { |
| return default_value; |
| } |
| -}; |
| +} |
| int32 GetTransportSendBufferSize(const base::DictionaryValue& options) { |
| // Socket send buffer size needs to be at least greater than one burst |
| @@ -183,17 +188,12 @@ void CastTransportSenderImpl::InitializeAudio( |
| return; |
| } |
| - audio_rtcp_session_.reset( |
| - new Rtcp(base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage, |
| - weak_factory_.GetWeakPtr(), config.ssrc, |
| - cast_message_cb), |
| - rtt_cb, |
| - base::Bind(&CastTransportSenderImpl::OnReceivedLogMessage, |
| - weak_factory_.GetWeakPtr(), AUDIO_EVENT), |
| - clock_, |
| - &pacer_, |
| - config.ssrc, |
| - config.feedback_ssrc)); |
| + audio_rtcp_session_.reset(new SenderRtcpSession( |
| + base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage, |
| + weak_factory_.GetWeakPtr(), config.ssrc, cast_message_cb), |
| + rtt_cb, base::Bind(&CastTransportSenderImpl::OnReceivedLogMessage, |
| + weak_factory_.GetWeakPtr(), AUDIO_EVENT), |
| + clock_, &pacer_, config.ssrc, config.feedback_ssrc)); |
| pacer_.RegisterAudioSsrc(config.ssrc); |
| AddValidSsrc(config.feedback_ssrc); |
| status_callback_.Run(TRANSPORT_AUDIO_INITIALIZED); |
| @@ -217,17 +217,12 @@ void CastTransportSenderImpl::InitializeVideo( |
| return; |
| } |
| - video_rtcp_session_.reset( |
| - new Rtcp(base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage, |
| - weak_factory_.GetWeakPtr(), config.ssrc, |
| - cast_message_cb), |
| - rtt_cb, |
| - base::Bind(&CastTransportSenderImpl::OnReceivedLogMessage, |
| - weak_factory_.GetWeakPtr(), VIDEO_EVENT), |
| - clock_, |
| - &pacer_, |
| - config.ssrc, |
| - config.feedback_ssrc)); |
| + video_rtcp_session_.reset(new SenderRtcpSession( |
| + base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage, |
| + weak_factory_.GetWeakPtr(), config.ssrc, cast_message_cb), |
| + rtt_cb, base::Bind(&CastTransportSenderImpl::OnReceivedLogMessage, |
| + weak_factory_.GetWeakPtr(), VIDEO_EVENT), |
| + clock_, &pacer_, config.ssrc, config.feedback_ssrc)); |
| pacer_.RegisterVideoSsrc(config.ssrc); |
| AddValidSsrc(config.feedback_ssrc); |
| status_callback_.Run(TRANSPORT_VIDEO_INITIALIZED); |
| @@ -270,11 +265,11 @@ void CastTransportSenderImpl::SendSenderReport( |
| base::TimeTicks current_time, |
| RtpTimeTicks current_time_as_rtp_timestamp) { |
| if (audio_sender_ && ssrc == audio_sender_->ssrc()) { |
| - audio_rtcp_session_->SendRtcpFromRtpSender( |
| + audio_rtcp_session_->SendRtcpReport( |
| current_time, current_time_as_rtp_timestamp, |
| audio_sender_->send_packet_count(), audio_sender_->send_octet_count()); |
| } else if (video_sender_ && ssrc == video_sender_->ssrc()) { |
| - video_rtcp_session_->SendRtcpFromRtpSender( |
| + video_rtcp_session_->SendRtcpReport( |
| current_time, current_time_as_rtp_timestamp, |
| video_sender_->send_packet_count(), video_sender_->send_octet_count()); |
| } else { |
| @@ -359,8 +354,8 @@ bool CastTransportSenderImpl::OnReceivedPacket(scoped_ptr<Packet> packet) { |
| const uint8_t* const data = &packet->front(); |
| const size_t length = packet->size(); |
| uint32 ssrc; |
| - if (Rtcp::IsRtcpPacket(data, length)) { |
| - ssrc = Rtcp::GetSsrcOfSender(data, length); |
| + if (IsRtcpPacket(data, length)) { |
| + ssrc = GetSsrcOfSender(data, length); |
| } else if (!RtpParser::ParseSsrc(data, length, &ssrc)) { |
| VLOG(1) << "Invalid RTP packet."; |
| return false; |
| @@ -477,18 +472,9 @@ void CastTransportSenderImpl::SendRtcpFromRtpReceiver( |
| base::TimeDelta target_delay, |
| const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events, |
| const RtpReceiverStatistics* rtp_receiver_statistics) { |
| - const Rtcp rtcp(RtcpCastMessageCallback(), |
| - RtcpRttCallback(), |
| - RtcpLogMessageCallback(), |
| - clock_, |
| - &pacer_, |
| - ssrc, |
| - sender_ssrc); |
| - rtcp.SendRtcpFromRtpReceiver(time_data, |
| - cast_message, |
| - target_delay, |
| - rtcp_events, |
| - rtp_receiver_statistics); |
| + const ReceiverRtcpSession rtcp(clock_, &pacer_, ssrc, sender_ssrc); |
| + rtcp.SendRtcpReport(time_data, cast_message, target_delay, rtcp_events, |
|
miu
2015/12/12 00:53:25
Huh. Interesting that this was in the CastTranspo
Irfan
2015/12/12 01:07:37
will do because this is indeed confusing code, but
|
| + rtp_receiver_statistics); |
| } |
| } // namespace cast |