Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 3c9daff..cc7ec5d 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -58,14 +58,15 @@
std::unique_ptr<voe::ChannelSendProxy> CreateChannelAndProxy(
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
+ MediaTransportInterface* media_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options) {
return absl::make_unique<voe::ChannelSendProxy>(
- absl::make_unique<voe::ChannelSend>(worker_queue, module_process_thread,
- rtcp_rtt_stats, event_log,
- frame_encryptor, crypto_options));
+ absl::make_unique<voe::ChannelSend>(
+ worker_queue, module_process_thread, media_transport, rtcp_rtt_stats,
+ event_log, frame_encryptor, crypto_options));
}
} // namespace
@@ -97,7 +98,7 @@
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
- RtpTransportControllerSendInterface* transport,
+ RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
@@ -106,7 +107,7 @@
: AudioSendStream(config,
audio_state,
worker_queue,
- transport,
+ rtp_transport,
bitrate_allocator,
event_log,
rtcp_rtt_stats,
@@ -114,6 +115,7 @@
overall_call_lifetime,
CreateChannelAndProxy(worker_queue,
module_process_thread,
+ config.media_transport,
rtcp_rtt_stats,
event_log,
config.frame_encryptor,
@@ -123,7 +125,7 @@
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
- RtpTransportControllerSendInterface* transport,
+ RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
@@ -131,12 +133,13 @@
TimeInterval* overall_call_lifetime,
std::unique_ptr<voe::ChannelSendProxy> channel_proxy)
: worker_queue_(worker_queue),
- config_(Config(nullptr)),
+ config_(Config(/*send_transport=*/nullptr,
+ /*media_transport=*/nullptr)),
audio_state_(audio_state),
channel_proxy_(std::move(channel_proxy)),
event_log_(event_log),
bitrate_allocator_(bitrate_allocator),
- transport_(transport),
+ rtp_transport_(rtp_transport),
packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
kPacketLossRateMinNumAckedPackets,
kRecoverablePacketLossRateMinNumAckedPairs),
@@ -148,7 +151,11 @@
RTC_DCHECK(audio_state_);
RTC_DCHECK(channel_proxy_);
RTC_DCHECK(bitrate_allocator_);
- RTC_DCHECK(transport);
+ // TODO(nisse): Eventually, we should have only media_transport. But for the
+ // time being, we can have either. When media transport is injected, there
+ // should be no rtp_transport, and below check should be strengthened to XOR
+ // (either rtp_transport or media_transport but not both).
+ RTC_DCHECK(rtp_transport || config.media_transport);
RTC_DCHECK(overall_call_lifetime_);
channel_proxy_->SetRTCPStatus(true);
@@ -158,17 +165,22 @@
ConfigureStream(this, config, true);
pacer_thread_checker_.DetachFromThread();
- // Signal congestion controller this object is ready for OnPacket* callbacks.
- transport_->RegisterPacketFeedbackObserver(this);
+ if (rtp_transport_) {
+ // Signal congestion controller this object is ready for OnPacket*
+ // callbacks.
+ rtp_transport_->RegisterPacketFeedbackObserver(this);
+ }
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
RTC_DCHECK(!sending_);
- transport_->DeRegisterPacketFeedbackObserver(this);
- channel_proxy_->RegisterTransport(nullptr);
- channel_proxy_->ResetSenderCongestionControlObjects();
+ if (rtp_transport_) {
+ rtp_transport_->DeRegisterPacketFeedbackObserver(this);
+ channel_proxy_->RegisterTransport(nullptr);
+ channel_proxy_->ResetSenderCongestionControlObjects();
+ }
// Lifetime can only be updated after deregistering
// |timed_send_transport_adapter_| in the underlying channel object to avoid
// data races in |active_lifetime_|.
@@ -272,14 +284,16 @@
// Probing in application limited region is only used in combination with
// send side congestion control, wich depends on feedback packets which
// requires transport sequence numbers to be enabled.
- stream->transport_->EnablePeriodicAlrProbing(true);
- bandwidth_observer = stream->transport_->GetBandwidthObserver();
+ if (stream->rtp_transport_) {
+ stream->rtp_transport_->EnablePeriodicAlrProbing(true);
+ bandwidth_observer = stream->rtp_transport_->GetBandwidthObserver();
+ }
}
-
- channel_proxy->RegisterSenderCongestionControlObjects(stream->transport_,
- bandwidth_observer);
+ if (stream->rtp_transport_) {
+ channel_proxy->RegisterSenderCongestionControlObjects(
+ stream->rtp_transport_, bandwidth_observer);
+ }
}
-
// MID RTP header extension.
if ((first_time || new_ids.mid != old_ids.mid ||
new_config.rtp.mid != old_config.rtp.mid) &&
@@ -312,7 +326,7 @@
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") ||
webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) {
// Audio BWE is enabled.
- transport_->packet_sender()->SetAccountForAudioPackets(true);
+ rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
config_.bitrate_priority,
@@ -725,13 +739,13 @@
!new_config.has_dscp &&
(has_transport_sequence_number ||
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
- stream->transport_->packet_sender()->SetAccountForAudioPackets(true);
+ stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
stream->ConfigureBitrateObserver(
new_config.min_bitrate_bps, new_config.max_bitrate_bps,
new_config.bitrate_priority, has_transport_sequence_number);
stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
- stream->transport_->packet_sender()->SetAccountForAudioPackets(false);
+ stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false);
stream->RemoveBitrateObserver();
stream->rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
