Introduce MediaTransportConfig
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 4646eb1..7539f37 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -123,6 +123,7 @@
deps = [
":audio",
":audio_end_to_end_test",
+ "../api:libjingle_peerconnection_api",
"../api:loopback_media_transport",
"../api:mock_audio_mixer",
"../api:mock_frame_decryptor",
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index b98c213..32617aa 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -56,7 +56,7 @@
ss << "{rtp: " << rtp.ToString();
ss << ", rtcp_send_transport: "
<< (rtcp_send_transport ? "(Transport)" : "null");
- ss << ", media_transport: " << (media_transport ? "(Transport)" : "null");
+ ss << ", media_transport_config: " << media_transport_config.DebugString();
if (!sync_group.empty()) {
ss << ", sync_group: " << sync_group;
}
@@ -77,7 +77,7 @@
static_cast<internal::AudioState*>(audio_state);
return voe::CreateChannelReceive(
clock, module_process_thread, internal_audio_state->audio_device_module(),
- config.media_transport, config.rtcp_send_transport, event_log,
+ config.media_transport_config, config.rtcp_send_transport, event_log,
config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
config.jitter_buffer_enable_rtx_handling, config.decoder_factory,
@@ -122,7 +122,7 @@
module_process_thread_checker_.Detach();
- if (!config.media_transport) {
+ if (!config.media_transport_config.media_transport) {
RTC_DCHECK(receiver_controller);
RTC_DCHECK(packet_router);
// Configure bandwidth estimation.
@@ -140,7 +140,7 @@
RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc;
Stop();
channel_receive_->SetAssociatedSendChannel(nullptr);
- if (!config_.media_transport) {
+ if (!config_.media_transport_config.media_transport) {
channel_receive_->ResetReceiverCongestionControlObjects();
}
}
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 303e0e8..b97217c 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -220,7 +220,8 @@
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
"{rtp_history_ms: 0}, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
- "rtcp_send_transport: null, media_transport: null}",
+ "rtcp_send_transport: null, media_transport_config: {media_transport: "
+ "null}}",
config.ToString());
}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index a72292d..942551b 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -21,6 +21,7 @@
#include "api/call/transport.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/function_view.h"
+#include "api/media_transport_config.h"
#include "audio/audio_state.h"
#include "audio/channel_send.h"
#include "audio/conversion.h"
@@ -104,7 +105,7 @@
voe::CreateChannelSend(clock,
task_queue_factory,
module_process_thread,
- config.media_transport,
+ config.media_transport_config,
/*overhead_observer=*/this,
config.send_transport,
rtcp_rtt_stats,
@@ -127,8 +128,7 @@
std::unique_ptr<voe::ChannelSendInterface> channel_send)
: clock_(clock),
worker_queue_(rtp_transport->GetWorkerQueue()),
- config_(Config(/*send_transport=*/nullptr,
- /*media_transport=*/nullptr)),
+ config_(Config(/*send_transport=*/nullptr, MediaTransportConfig())),
audio_state_(audio_state),
channel_send_(std::move(channel_send)),
event_log_(event_log),
@@ -151,15 +151,15 @@
// 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);
- if (config.media_transport) {
+ RTC_DCHECK(rtp_transport || config.media_transport_config.media_transport);
+ if (config.media_transport_config.media_transport) {
// TODO(sukhanov): Currently media transport audio overhead is considered
// constant, we will not get overhead_observer calls when using
// media_transport. In the future when we introduce RTP media transport we
// should make audio overhead interface consistent and work for both RTP and
// non-RTP implementations.
audio_overhead_per_packet_bytes_ =
- config.media_transport->GetAudioPacketOverhead();
+ config.media_transport_config.media_transport->GetAudioPacketOverhead();
}
rtp_rtcp_module_ = channel_send_->GetRtpRtcp();
RTC_DCHECK(rtp_rtcp_module_);
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 5ddc5e1..4531755 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -136,7 +136,7 @@
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
: clock_(1000000),
task_queue_factory_(CreateDefaultTaskQueueFactory()),
- stream_config_(/*send_transport=*/nullptr, /*media_transport=*/nullptr),
+ stream_config_(/*send_transport=*/nullptr, MediaTransportConfig()),
audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
bitrate_allocator_(&clock_, &limit_observer_),
worker_queue_(task_queue_factory_->CreateTaskQueue(
@@ -321,7 +321,7 @@
TEST(AudioSendStreamTest, ConfigToString) {
AudioSendStream::Config config(/*send_transport=*/nullptr,
- /*media_transport=*/nullptr);
+ MediaTransportConfig());
config.rtp.ssrc = kSsrc;
config.rtp.c_name = kCName;
config.min_bitrate_bps = 12000;
@@ -340,7 +340,7 @@
"{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
"c_name: foo_name}, rtcp_report_interval_ms: 2500, "
- "send_transport: null, media_transport: null, "
+ "send_transport: null, media_transport_config: {media_transport: null}, "
"min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"cng_payload_type: 42, payload_type: 103, "
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index f65d125..85a029d 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -79,7 +79,7 @@
ChannelReceive(Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t remote_ssrc,
@@ -157,6 +157,12 @@
std::vector<RtpSource> GetSources() const override;
+ // TODO(sukhanov): Return const pointer. It requires making media transport
+ // getters like GetLatestTargetTransferRate to be also const.
+ MediaTransportInterface* media_transport() const {
+ return media_transport_config_.media_transport;
+ }
+
private:
bool ReceivePacket(const uint8_t* packet,
size_t packet_length,
@@ -254,7 +260,7 @@
rtc::ThreadChecker construction_thread_;
- MediaTransportInterface* const media_transport_;
+ MediaTransportConfig media_transport_config_;
// E2EE Audio Frame Decryption
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
@@ -265,7 +271,7 @@
size_t payloadSize,
const RTPHeader& rtp_header) {
// We should not be receiving any RTP packets if media_transport is set.
- RTC_CHECK(!media_transport_);
+ RTC_CHECK(!media_transport());
if (!Playing()) {
// Avoid inserting into NetEQ when we are not playing. Count the
@@ -296,7 +302,7 @@
// MediaTransportAudioSinkInterface override.
void ChannelReceive::OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) {
- RTC_CHECK(media_transport_);
+ RTC_CHECK(media_transport());
if (!Playing()) {
// Avoid inserting into NetEQ when we are not playing. Count the
@@ -432,7 +438,7 @@
Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t remote_ssrc,
@@ -458,7 +464,7 @@
_audioDeviceModulePtr(audio_device_module),
_outputGain(1.0f),
associated_send_channel_(nullptr),
- media_transport_(media_transport),
+ media_transport_config_(media_transport_config),
frame_decryptor_(frame_decryptor),
crypto_options_(crypto_options) {
// TODO(nisse): Use _moduleProcessThreadPtr instead?
@@ -503,16 +509,16 @@
// RTCP is enabled by default.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
- if (media_transport_) {
- media_transport_->SetReceiveAudioSink(this);
+ if (media_transport()) {
+ media_transport()->SetReceiveAudioSink(this);
}
}
ChannelReceive::~ChannelReceive() {
RTC_DCHECK(construction_thread_.IsCurrent());
- if (media_transport_) {
- media_transport_->SetReceiveAudioSink(nullptr);
+ if (media_transport()) {
+ media_transport()->SetReceiveAudioSink(nullptr);
}
StopPlayout();
@@ -921,8 +927,8 @@
}
int64_t ChannelReceive::GetRTT() const {
- if (media_transport_) {
- auto target_rate = media_transport_->GetLatestTargetTransferRate();
+ if (media_transport()) {
+ auto target_rate = media_transport()->GetLatestTargetTransferRate();
if (target_rate.has_value()) {
return target_rate->network_estimate.round_trip_time.ms();
}
@@ -966,7 +972,7 @@
Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t remote_ssrc,
@@ -979,7 +985,7 @@
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options) {
return absl::make_unique<ChannelReceive>(
- clock, module_process_thread, audio_device_module, media_transport,
+ clock, module_process_thread, audio_device_module, media_transport_config,
rtcp_send_transport, rtc_event_log, remote_ssrc,
jitter_buffer_max_packets, jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling,
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 1f78874..d29f624 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -22,6 +22,7 @@
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
+#include "api/media_transport_config.h"
#include "api/media_transport_interface.h"
#include "api/rtp_receiver_interface.h"
#include "call/rtp_packet_sink_interface.h"
@@ -143,7 +144,7 @@
Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t remote_ssrc,
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index e8360cb..38e89d8 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -89,7 +89,7 @@
ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
@@ -205,7 +205,9 @@
RTC_RUN_ON(encoder_queue_);
// Return media transport or nullptr if using RTP.
- MediaTransportInterface* media_transport() { return media_transport_; }
+ MediaTransportInterface* media_transport() {
+ return media_transport_config_.media_transport;
+ }
// Called on the encoder task queue when a new input audio frame is ready
// for encoding.
@@ -266,7 +268,7 @@
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
- MediaTransportInterface* const media_transport_;
+ MediaTransportConfig media_transport_config_;
int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
rtc::CriticalSection media_transport_lock_;
@@ -618,7 +620,7 @@
ChannelSend::ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
@@ -642,7 +644,7 @@
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
- media_transport_(media_transport),
+ media_transport_config_(media_transport_config),
frame_encryptor_(frame_encryptor),
crypto_options_(crypto_options),
encoder_queue_(task_queue_factory->CreateTaskQueue(
@@ -659,7 +661,7 @@
// transport. All of this logic should be moved to the future
// RTPMediaTransport. In this case it means that overhead and bandwidth
// observers should not be called when using media transport.
- if (!media_transport_) {
+ if (!media_transport_config.media_transport) {
configuration.overhead_observer = overhead_observer;
configuration.bandwidth_callback = rtcp_observer_.get();
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
@@ -689,10 +691,11 @@
// We want to invoke the 'TargetRateObserver' and |OnOverheadChanged|
// callbacks after the audio_coding_ is fully initialized.
- if (media_transport_) {
+ if (media_transport_config.media_transport) {
RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers.";
- media_transport_->AddTargetTransferRateObserver(this);
- media_transport_->SetAudioOverheadObserver(overhead_observer);
+ media_transport_config.media_transport->AddTargetTransferRateObserver(this);
+ media_transport_config.media_transport->SetAudioOverheadObserver(
+ overhead_observer);
} else {
RTC_DLOG(LS_INFO) << "Not setting media_transport_ rate observers.";
}
@@ -714,9 +717,10 @@
ChannelSend::~ChannelSend() {
RTC_DCHECK(construction_thread_.IsCurrent());
- if (media_transport_) {
- media_transport_->RemoveTargetTransferRateObserver(this);
- media_transport_->SetAudioOverheadObserver(nullptr);
+ if (media_transport_config_.media_transport) {
+ media_transport_config_.media_transport->RemoveTargetTransferRateObserver(
+ this);
+ media_transport_config_.media_transport->SetAudioOverheadObserver(nullptr);
}
StopSend();
@@ -779,7 +783,7 @@
encoder->RtpTimestampRateHz(),
encoder->NumChannels(), 0);
- if (media_transport_) {
+ if (media_transport_config_.media_transport) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_payload_type_ = payload_type;
// TODO(nisse): Currently broken for G722, since timestamps passed through
@@ -856,7 +860,7 @@
void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// May be called on either worker thread or network thread.
- if (media_transport_) {
+ if (media_transport_config_.media_transport) {
// Ignore RTCP packets while media transport is used.
// Those packets should not arrive, but we are seeing occasional packets.
return;
@@ -931,7 +935,7 @@
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(!sending_);
- if (media_transport_) {
+ if (media_transport_config_.media_transport) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_channel_id_ = ssrc;
}
@@ -1165,12 +1169,13 @@
}
int64_t ChannelSend::GetRTT() const {
- if (media_transport_) {
+ if (media_transport_config_.media_transport) {
// GetRTT is generally used in the RTCP codepath, where media transport is
// not present and so it shouldn't be needed. But it's also invoked in
// 'GetStats' method, and for now returning media transport RTT here gives
// us "free" rtt stats for media transport.
- auto target_rate = media_transport_->GetLatestTargetTransferRate();
+ auto target_rate =
+ media_transport_config_.media_transport->GetLatestTargetTransferRate();
if (target_rate.has_value()) {
return target_rate.value().network_estimate.round_trip_time.ms();
}
@@ -1214,7 +1219,7 @@
// AudioSendStream. Since AudioSendStream owns encoder and configures ANA, it
// makes sense to consolidate all rate (and overhead) calculation there.
void ChannelSend::OnTargetTransferRate(TargetTransferRate rate) {
- RTC_DCHECK(media_transport_);
+ RTC_DCHECK(media_transport_config_.media_transport);
OnReceivedRtt(rate.network_estimate.round_trip_time.ms());
}
@@ -1230,7 +1235,7 @@
Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
@@ -1240,7 +1245,7 @@
bool extmap_allow_mixed,
int rtcp_report_interval_ms) {
return absl::make_unique<ChannelSend>(
- clock, task_queue_factory, module_process_thread, media_transport,
+ clock, task_queue_factory, module_process_thread, media_transport_config,
overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,
rtcp_report_interval_ms);
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 45f7b1e..fb98be3 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -19,6 +19,7 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/crypto/crypto_options.h"
#include "api/function_view.h"
+#include "api/media_transport_config.h"
#include "api/media_transport_interface.h"
#include "api/task_queue/task_queue_factory.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -125,7 +126,7 @@
Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
diff --git a/audio/test/media_transport_test.cc b/audio/test/media_transport_test.cc
index 7794b43..4594c3e 100644
--- a/audio/test/media_transport_test.cc
+++ b/audio/test/media_transport_test.cc
@@ -13,6 +13,7 @@
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "api/media_transport_config.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/loopback_media_transport.h"
#include "api/test/mock_audio_mixer.h"
@@ -100,7 +101,8 @@
// TODO(nisse): Update AudioReceiveStream to not require rtcp_send_transport
// when a MediaTransport is provided.
receive_config.rtcp_send_transport = &rtcp_send_transport;
- receive_config.media_transport = transport_pair.first();
+ receive_config.media_transport_config.media_transport =
+ transport_pair.first();
receive_config.decoder_map.emplace(kPayloadTypeOpus, audio_format);
receive_config.decoder_factory =
CreateAudioDecoderFactory<AudioDecoderOpus>();
@@ -116,7 +118,8 @@
// TODO(nisse): Update AudioSendStream to not require send_transport when a
// MediaTransport is provided.
- AudioSendStream::Config send_config(&send_transport, transport_pair.second());
+ AudioSendStream::Config send_config(
+ &send_transport, webrtc::MediaTransportConfig(transport_pair.second()));
send_config.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kPayloadTypeOpus, audio_format);
send_config.encoder_factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
