Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index f31e11479bfd1ae7d641e8e791d626cbac53d824..576ee1dfb35a1c30b1386cb866b6afe8982c90a9 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -34,7 +34,7 @@ |
#include "webrtc/call/bitrate_allocator.h" |
#include "webrtc/call/call.h" |
#include "webrtc/call/flexfec_receive_stream_impl.h" |
-#include "webrtc/call/rtp_demuxer.h" |
+#include "webrtc/call/rtp_stream_receiver_controller.h" |
#include "webrtc/call/rtp_transport_controller_send.h" |
#include "webrtc/config.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
@@ -277,10 +277,10 @@ class Call : public webrtc::Call, |
std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ |
GUARDED_BY(receive_crit_); |
- // TODO(nisse): Should eventually be part of injected |
- // RtpTransportControllerReceive, with a single demuxer in the bundled case. |
- RtpDemuxer audio_rtp_demuxer_ GUARDED_BY(receive_crit_); |
- RtpDemuxer video_rtp_demuxer_ GUARDED_BY(receive_crit_); |
+ // TODO(nisse): Should eventually be injected at creation, |
+ // with a single object in the bundled case. |
+ RtpStreamReceiverController audio_receiver_controller; |
+ RtpStreamReceiverController video_receiver_controller; |
// This extra map is used for receive processing which is |
// independent of media type. |
@@ -648,12 +648,11 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( |
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); |
RTC_DCHECK_RUN_ON(&configuration_thread_checker_); |
event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config)); |
- AudioReceiveStream* receive_stream = |
- new AudioReceiveStream(transport_send_->packet_router(), config, |
- config_.audio_state, event_log_); |
+ AudioReceiveStream* receive_stream = new AudioReceiveStream( |
+ &audio_receiver_controller, transport_send_->packet_router(), config, |
+ config_.audio_state, event_log_); |
{ |
WriteLockScoped write_lock(*receive_crit_); |
- audio_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream); |
receive_rtp_config_[config.rtp.remote_ssrc] = |
ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config)); |
audio_receive_streams_.insert(receive_stream); |
@@ -685,8 +684,6 @@ void Call::DestroyAudioReceiveStream( |
uint32_t ssrc = config.rtp.remote_ssrc; |
receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config)) |
->RemoveStream(ssrc); |
- size_t num_deleted = audio_rtp_demuxer_.RemoveSink(audio_receive_stream); |
- RTC_DCHECK(num_deleted == 1); |
audio_receive_streams_.erase(audio_receive_stream); |
const std::string& sync_group = audio_receive_stream->config().sync_group; |
const auto it = sync_stream_mapping_.find(sync_group); |
@@ -778,19 +775,17 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); |
RTC_DCHECK_RUN_ON(&configuration_thread_checker_); |
- VideoReceiveStream* receive_stream = |
- new VideoReceiveStream(num_cpu_cores_, transport_send_->packet_router(), |
- std::move(configuration), |
- module_process_thread_.get(), call_stats_.get()); |
+ VideoReceiveStream* receive_stream = new VideoReceiveStream( |
+ &video_receiver_controller, num_cpu_cores_, |
+ transport_send_->packet_router(), std::move(configuration), |
+ module_process_thread_.get(), call_stats_.get()); |
const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); |
ReceiveRtpConfig receive_config(config.rtp.extensions, |
UseSendSideBwe(config)); |
{ |
WriteLockScoped write_lock(*receive_crit_); |
- video_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream); |
if (config.rtp.rtx_ssrc) { |
- video_rtp_demuxer_.AddSink(config.rtp.rtx_ssrc, receive_stream); |
// We record identical config for the rtx stream as for the main |
// stream. Since the transport_send_cc negotiation is per payload |
// type, we may get an incorrect value for the rtx stream, but |
@@ -819,8 +814,6 @@ void Call::DestroyVideoReceiveStream( |
WriteLockScoped write_lock(*receive_crit_); |
// Remove all ssrcs pointing to a receive stream. As RTX retransmits on a |
// separate SSRC there can be either one or two. |
- size_t num_deleted = video_rtp_demuxer_.RemoveSink(receive_stream_impl); |
- RTC_DCHECK_GE(num_deleted, 1); |
receive_rtp_config_.erase(config.rtp.remote_ssrc); |
if (config.rtp.rtx_ssrc) { |
receive_rtp_config_.erase(config.rtp.rtx_ssrc); |
@@ -842,17 +835,12 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream( |
RTC_DCHECK_RUN_ON(&configuration_thread_checker_); |
RecoveredPacketReceiver* recovered_packet_receiver = this; |
- FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl( |
- config, recovered_packet_receiver, call_stats_->rtcp_rtt_stats(), |
- module_process_thread_.get()); |
+ FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl( |
+ &video_receiver_controller, config, recovered_packet_receiver, |
+ call_stats_->rtcp_rtt_stats(), module_process_thread_.get()); |
{ |
WriteLockScoped write_lock(*receive_crit_); |
- video_rtp_demuxer_.AddSink(config.remote_ssrc, receive_stream); |
- |
- for (auto ssrc : config.protected_media_ssrcs) |
- video_rtp_demuxer_.AddSink(ssrc, receive_stream); |
- |
RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) == |
receive_rtp_config_.end()); |
receive_rtp_config_[config.remote_ssrc] = |
@@ -883,7 +871,6 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) { |
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be |
// destroyed. |
- video_rtp_demuxer_.RemoveSink(receive_stream_impl); |
receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config)) |
->RemoveStream(ssrc); |
} |
@@ -1321,14 +1308,14 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
NotifyBweOfReceivedPacket(*parsed_packet, media_type); |
if (media_type == MediaType::AUDIO) { |
- if (audio_rtp_demuxer_.OnRtpPacket(*parsed_packet)) { |
+ if (audio_receiver_controller.OnRtpPacket(*parsed_packet)) { |
received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); |
event_log_->LogRtpHeader(kIncomingPacket, packet, length); |
return DELIVERY_OK; |
} |
} else if (media_type == MediaType::VIDEO) { |
- if (video_rtp_demuxer_.OnRtpPacket(*parsed_packet)) { |
+ if (video_receiver_controller.OnRtpPacket(*parsed_packet)) { |
received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
event_log_->LogRtpHeader(kIncomingPacket, packet, length); |
@@ -1364,7 +1351,7 @@ void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) { |
parsed_packet->set_recovered(true); |
- video_rtp_demuxer_.OnRtpPacket(*parsed_packet); |
+ video_receiver_controller.OnRtpPacket(*parsed_packet); |
} |
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet, |