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Unified Diff: media/cast/net/cast_transport_sender_impl.cc

Issue 1520613004: cast: Split Rtcp into two for sender and receiver (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@timestamp
Patch Set: Add missing rtcp_session.h and a few comments Created 5 years ago
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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

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