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Move channel creation functions into RtpTransceiver

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This breaks the link from sdp_offer_answer.cc to channel.h.

Bug: webrtc:13931
Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36757}
This commit is contained in:
Harald Alvestrand 2022-05-04 10:32:30 +00:00 committed by WebRTC LUCI CQ
parent 1bcd827e9b
commit 8f42992787
15 changed files with 294 additions and 93 deletions
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20
pc/BUILD.gn
View File

@ -1009,12 +1009,26 @@ rtc_source_set("data_channel_controller") {
]
}
rtc_source_set("peer_connection_sdp_methods") {
visibility = [ ":*" ]
sources = [ "peer_connection_sdp_methods.h" ]
deps = [
":jsep_transport_controller",
":peer_connection_message_handler",
":sctp_data_channel",
":usage_pattern",
"../api:libjingle_peerconnection_api",
"../call:call_interfaces",
]
}
rtc_source_set("peer_connection_internal") {
visibility = [ ":*" ]
sources = [ "peer_connection_internal.h" ]
deps = [
":jsep_transport_controller",
":peer_connection_message_handler",
":peer_connection_sdp_methods",
":rtp_transceiver",
":rtp_transmission_manager",
":sctp_data_channel",
@ -1022,6 +1036,7 @@ rtc_source_set("peer_connection_internal") {
"../call:call_interfaces",
]
}
rtc_source_set("rtc_stats_collector") {
visibility = [
":*",
@ -1133,6 +1148,7 @@ rtc_source_set("sdp_offer_answer") {
":webrtc_session_description_factory",
"../api:array_view",
"../api:audio_options_api",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
@ -1579,8 +1595,10 @@ rtc_library("rtp_transceiver") {
"rtp_transceiver.h",
]
deps = [
":channel",
":channel_interface",
":channel_manager",
":peer_connection_sdp_methods",
":proxy",
":rtp_media_utils",
":rtp_parameters_conversion",
@ -1591,6 +1609,7 @@ rtc_library("rtp_transceiver") {
":rtp_transport_internal",
":session_description",
"../api:array_view",
"../api:audio_options_api",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
@ -1598,6 +1617,7 @@ rtc_library("rtp_transceiver") {
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue",
"../api/video:video_bitrate_allocator_factory",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:location",

6
pc/channel.cc
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@ -117,7 +117,7 @@ BaseChannel::BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
@ -818,7 +818,7 @@ VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> media_channel,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
@ -941,7 +941,7 @@ VideoChannel::VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)

6
pc/channel.h
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@ -83,7 +83,7 @@ class BaseChannel : public ChannelInterface,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
@ -361,7 +361,7 @@ class VoiceChannel : public BaseChannel {
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> channel,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
@ -407,7 +407,7 @@ class VideoChannel : public BaseChannel {
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);

4
pc/channel_factory_interface.h
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@ -31,7 +31,7 @@ class ChannelFactoryInterface {
virtual std::unique_ptr<VideoChannel> CreateVideoChannel(
webrtc::Call* call,
const MediaConfig& media_config,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
const VideoOptions& options,
@ -41,7 +41,7 @@ class ChannelFactoryInterface {
virtual std::unique_ptr<VoiceChannel> CreateVoiceChannel(
webrtc::Call* call,
const MediaConfig& media_config,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
const AudioOptions& options) = 0;

4
pc/channel_manager.cc
View File

@ -152,7 +152,7 @@ ChannelManager::GetSupportedVideoRtpHeaderExtensions() const {
std::unique_ptr<VoiceChannel> ChannelManager::CreateVoiceChannel(
webrtc::Call* call,
const MediaConfig& media_config,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
const AudioOptions& options) {
@ -188,7 +188,7 @@ std::unique_ptr<VoiceChannel> ChannelManager::CreateVoiceChannel(
std::unique_ptr<VideoChannel> ChannelManager::CreateVideoChannel(
webrtc::Call* call,
const MediaConfig& media_config,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
const VideoOptions& options,

4
pc/channel_manager.h
View File

@ -84,7 +84,7 @@ class ChannelManager : public ChannelFactoryInterface {
std::unique_ptr<VoiceChannel> CreateVoiceChannel(
webrtc::Call* call,
const MediaConfig& media_config,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
const AudioOptions& options) override;
@ -95,7 +95,7 @@ class ChannelManager : public ChannelFactoryInterface {
std::unique_ptr<VideoChannel> CreateVideoChannel(
webrtc::Call* call,
const MediaConfig& media_config,
const std::string& mid,
absl::string_view mid,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
const VideoOptions& options,

16
pc/jsep_transport_collection.cc
View File

@ -219,6 +219,22 @@ const cricket::JsepTransport* JsepTransportCollection::GetTransportForMid(
return it == mid_to_transport_.end() ? nullptr : it->second;
}
cricket::JsepTransport* JsepTransportCollection::GetTransportForMid(
absl::string_view mid) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// TODO(hta): should be a better way.
auto it = mid_to_transport_.find(std::string(mid));
return it == mid_to_transport_.end() ? nullptr : it->second;
}
const cricket::JsepTransport* JsepTransportCollection::GetTransportForMid(
absl::string_view mid) const {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// TODO(hta): Should be a better way
auto it = mid_to_transport_.find(std::string(mid));
return it == mid_to_transport_.end() ? nullptr : it->second;
}
bool JsepTransportCollection::SetTransportForMid(
const std::string& mid,
cricket::JsepTransport* jsep_transport) {

2
pc/jsep_transport_collection.h
View File

@ -118,6 +118,8 @@ class JsepTransportCollection {
cricket::JsepTransport* GetTransportForMid(const std::string& mid);
const cricket::JsepTransport* GetTransportForMid(
const std::string& mid) const;
cricket::JsepTransport* GetTransportForMid(absl::string_view mid);
const cricket::JsepTransport* GetTransportForMid(absl::string_view mid) const;
// Set transport for a MID. This may destroy a transport if it is no
// longer in use.
bool SetTransportForMid(const std::string& mid,

11
pc/jsep_transport_controller.cc
View File

@ -107,7 +107,7 @@ RTCError JsepTransportController::SetRemoteDescription(
}
RtpTransportInternal* JsepTransportController::GetRtpTransport(
const std::string& mid) const {
absl::string_view mid) const {
RTC_DCHECK_RUN_ON(network_thread_);
auto jsep_transport = GetJsepTransportForMid(mid);
if (!jsep_transport) {
@ -1000,6 +1000,15 @@ cricket::JsepTransport* JsepTransportController::GetJsepTransportForMid(
const std::string& mid) {
return transports_.GetTransportForMid(mid);
}
const cricket::JsepTransport* JsepTransportController::GetJsepTransportForMid(
absl::string_view mid) const {
return transports_.GetTransportForMid(mid);
}
cricket::JsepTransport* JsepTransportController::GetJsepTransportForMid(
absl::string_view mid) {
return transports_.GetTransportForMid(mid);
}
const cricket::JsepTransport* JsepTransportController::GetJsepTransportByName(
const std::string& transport_name) const {

6
pc/jsep_transport_controller.h
View File

@ -167,7 +167,7 @@ class JsepTransportController : public sigslot::has_slots<> {
// Get transports to be used for the provided `mid`. If bundling is enabled,
// calling GetRtpTransport for multiple MIDs may yield the same object.
RtpTransportInternal* GetRtpTransport(const std::string& mid) const;
RtpTransportInternal* GetRtpTransport(absl::string_view mid) const;
cricket::DtlsTransportInternal* GetDtlsTransport(const std::string& mid);
const cricket::DtlsTransportInternal* GetRtcpDtlsTransport(
const std::string& mid) const;
@ -362,6 +362,10 @@ class JsepTransportController : public sigslot::has_slots<> {
const std::string& mid) const RTC_RUN_ON(network_thread_);
cricket::JsepTransport* GetJsepTransportForMid(const std::string& mid)
RTC_RUN_ON(network_thread_);
const cricket::JsepTransport* GetJsepTransportForMid(
absl::string_view mid) const RTC_RUN_ON(network_thread_);
cricket::JsepTransport* GetJsepTransportForMid(absl::string_view mid)
RTC_RUN_ON(network_thread_);
// Get the JsepTransport without considering the BUNDLE group. Return nullptr
// if the JsepTransport is destroyed.

131
pc/peer_connection_sdp_methods.h Normal file
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@ -0,0 +1,131 @@
/*
* Copyright 2022 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef PC_PEER_CONNECTION_SDP_METHODS_H_
#define PC_PEER_CONNECTION_SDP_METHODS_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "api/peer_connection_interface.h"
#include "pc/jsep_transport_controller.h"
#include "pc/peer_connection_message_handler.h"
#include "pc/sctp_data_channel.h"
#include "pc/usage_pattern.h"
namespace webrtc {
class DataChannelController;
class RtpTransmissionManager;
class StatsCollector;
// This interface defines the functions that are needed for
// SdpOfferAnswerHandler to access PeerConnection internal state.
class PeerConnectionSdpMethods {
public:
virtual ~PeerConnectionSdpMethods() = default;
// The SDP session ID as defined by RFC 3264.
virtual std::string session_id() const = 0;
// Returns true if the ICE restart flag above was set, and no ICE restart has
// occurred yet for this transport (by applying a local description with
// changed ufrag/password). If the transport has been deleted as a result of
// bundling, returns false.
virtual bool NeedsIceRestart(const std::string& content_name) const = 0;
virtual absl::optional<std::string> sctp_mid() const = 0;
// Functions below this comment are known to only be accessed
// from SdpOfferAnswerHandler.
// Return a pointer to the active configuration.
virtual const PeerConnectionInterface::RTCConfiguration* configuration()
const = 0;
// Report the UMA metric SdpFormatReceived for the given remote description.
virtual void ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_description) = 0;
// Report the UMA metric BundleUsage for the given remote description.
virtual void ReportSdpBundleUsage(
const SessionDescriptionInterface& remote_description) = 0;
virtual PeerConnectionMessageHandler* message_handler() = 0;
virtual RtpTransmissionManager* rtp_manager() = 0;
virtual const RtpTransmissionManager* rtp_manager() const = 0;
virtual bool dtls_enabled() const = 0;
virtual const PeerConnectionFactoryInterface::Options* options() const = 0;
// Returns the CryptoOptions for this PeerConnection. This will always
// return the RTCConfiguration.crypto_options if set and will only default
// back to the PeerConnectionFactory settings if nothing was set.
virtual CryptoOptions GetCryptoOptions() = 0;
virtual JsepTransportController* transport_controller_s() = 0;
virtual JsepTransportController* transport_controller_n() = 0;
virtual DataChannelController* data_channel_controller() = 0;
virtual cricket::PortAllocator* port_allocator() = 0;
virtual StatsCollector* stats() = 0;
// Returns the observer. Will crash on CHECK if the observer is removed.
virtual PeerConnectionObserver* Observer() const = 0;
virtual bool GetSctpSslRole(rtc::SSLRole* role) = 0;
virtual PeerConnectionInterface::IceConnectionState
ice_connection_state_internal() = 0;
virtual void SetIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state) = 0;
virtual void NoteUsageEvent(UsageEvent event) = 0;
virtual bool IsClosed() const = 0;
// Returns true if the PeerConnection is configured to use Unified Plan
// semantics for creating offers/answers and setting local/remote
// descriptions. If this is true the RtpTransceiver API will also be available
// to the user. If this is false, Plan B semantics are assumed.
// TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
// sufficient time has passed.
virtual bool IsUnifiedPlan() const = 0;
virtual bool ValidateBundleSettings(
const cricket::SessionDescription* desc,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) = 0;
virtual absl::optional<std::string> GetDataMid() const = 0;
// Internal implementation for AddTransceiver family of methods. If
// `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool fire_callback = true) = 0;
// Asynchronously calls SctpTransport::Start() on the network thread for
// `sctp_mid()` if set. Called as part of setting the local description.
virtual void StartSctpTransport(int local_port,
int remote_port,
int max_message_size) = 0;
// Asynchronously adds a remote candidate on the network thread.
virtual void AddRemoteCandidate(const std::string& mid,
const cricket::Candidate& candidate) = 0;
virtual Call* call_ptr() = 0;
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
// this session.
virtual bool SrtpRequired() const = 0;
virtual bool SetupDataChannelTransport_n(const std::string& mid) = 0;
virtual void TeardownDataChannelTransport_n() = 0;
virtual void SetSctpDataMid(const std::string& mid) = 0;
virtual void ResetSctpDataMid() = 0;
virtual const FieldTrialsView& trials() const = 0;
};
} // namespace webrtc
#endif // PC_PEER_CONNECTION_SDP_METHODS_H_

48
pc/rtp_transceiver.cc
View File

@ -17,10 +17,12 @@
#include <vector>
#include "absl/algorithm/container.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"
#include "pc/channel.h"
#include "pc/channel_manager.h"
#include "pc/rtp_media_utils.h"
#include "pc/session_description.h"
@ -157,7 +159,51 @@ RtpTransceiver::~RtpTransceiver() {
StopInternal();
}
RTC_CHECK(!channel_) << "Missing call to SetChannel(nullptr)?";
RTC_CHECK(!channel_) << "Missing call to ClearChannel?";
}
RTCError RtpTransceiver::CreateChannel(
absl::string_view mid,
Call* call_ptr,
const cricket::MediaConfig& media_config,
bool srtp_required,
CryptoOptions crypto_options,
const cricket::AudioOptions& audio_options,
const cricket::VideoOptions& video_options,
VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
std::function<RtpTransportInternal*(absl::string_view)> transport_lookup) {
RTC_DCHECK_RUN_ON(thread_);
if (!channel_manager_->media_engine()) {
// TODO(hta): Must be a better way
return RTCError(RTCErrorType::INTERNAL_ERROR,
"No media engine for mid=" + std::string(mid));
}
std::unique_ptr<cricket::ChannelInterface> new_channel;
if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
// the worker thread. We shouldn't be using the `call_ptr_` hack here but
// simply be on the worker thread and use `call_` (update upstream code).
new_channel = channel_manager_->CreateVoiceChannel(
call_ptr, media_config, mid, srtp_required, crypto_options,
audio_options);
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type());
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
// the worker thread. We shouldn't be using the `call_ptr_` hack here but
// simply be on the worker thread and use `call_` (update upstream code).
new_channel = channel_manager_->CreateVideoChannel(
call_ptr, media_config, mid, srtp_required, crypto_options,
video_options, video_bitrate_allocator_factory);
}
if (!new_channel) {
// TODO(hta): Must be a better way
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to create channel for mid=" + std::string(mid));
}
SetChannel(std::move(new_channel), transport_lookup);
return RTCError::OK();
}
void RtpTransceiver::SetChannel(

23
pc/rtp_transceiver.h
View File

@ -20,6 +20,7 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_options.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
@ -30,6 +31,8 @@
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "media/base/media_channel.h"
#include "pc/channel_interface.h"
#include "pc/channel_manager.h"
#include "pc/proxy.h"
@ -45,6 +48,8 @@
namespace webrtc {
class PeerConnectionSdpMethods;
// Implementation of the public RtpTransceiverInterface.
//
// The RtpTransceiverInterface is only intended to be used with a PeerConnection
@ -66,10 +71,8 @@ namespace webrtc {
// with this m= section. Since the transceiver, senders, and receivers are
// reference counted and can be referenced from JavaScript (in Chromium), these
// objects must be ready to live for an arbitrary amount of time. The
// BaseChannel is not reference counted and is owned by the ChannelManager, so
// the PeerConnection must take care of creating/deleting the BaseChannel and
// setting the channel reference in the transceiver to null when it has been
// deleted.
// BaseChannel is not reference counted, so
// the PeerConnection must take care of creating/deleting the BaseChannel.
//
// The RtpTransceiver is specialized to either audio or video according to the
// MediaType specified in the constructor. Audio RtpTransceivers will have
@ -102,6 +105,18 @@ class RtpTransceiver : public RtpTransceiverInterface,
// the transceiver is not in the currently set local/remote description.
cricket::ChannelInterface* channel() const { return channel_.get(); }
// Creates the Voice/VideoChannel and sets it.
RTCError CreateChannel(
absl::string_view mid,
Call* call_ptr,
const cricket::MediaConfig& media_config,
bool srtp_required,
CryptoOptions crypto_options,
const cricket::AudioOptions& audio_options,
const cricket::VideoOptions& video_options,
VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
std::function<RtpTransportInternal*(absl::string_view)> transport_lookup);
// Sets the Voice/VideoChannel. The caller must pass in the correct channel
// implementation based on the type of the transceiver. The call must
// furthermore be made on the signaling thread.

101
pc/sdp_offer_answer.cc
View File

@ -26,6 +26,7 @@
#include "api/array_view.h"
#include "api/crypto/crypto_options.h"
#include "api/dtls_transport_interface.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
@ -39,7 +40,6 @@
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_description_factory.h"
#include "p2p/base/transport_info.h"
#include "pc/channel.h"
#include "pc/channel_interface.h"
#include "pc/dtls_transport.h"
#include "pc/media_stream.h"
@ -3652,24 +3652,17 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiverChannel(
}
} else {
if (!channel) {
std::unique_ptr<cricket::ChannelInterface> new_channel;
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
new_channel = CreateVoiceChannel(content.name);
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
new_channel = CreateVideoChannel(content.name);
}
if (!new_channel) {
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to create channel for mid=" + content.name);
}
// Note: this is a thread hop; the lambda will be executed
// on the network thread.
transceiver->internal()->SetChannel(
std::move(new_channel), [&](const std::string& mid) {
auto error = transceiver->internal()->CreateChannel(
content.name, pc_->call_ptr(), pc_->configuration()->media_config,
pc_->SrtpRequired(), pc_->GetCryptoOptions(), audio_options(),
video_options(), video_bitrate_allocator_factory_.get(),
[&](absl::string_view mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
if (!error.ok()) {
return error;
}
}
}
return RTCError::OK();
@ -4842,33 +4835,36 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
if (voice && !voice->rejected &&
!rtp_manager()->GetAudioTransceiver()->internal()->channel()) {
std::unique_ptr<cricket::VoiceChannel> voice_channel =
CreateVoiceChannel(voice->name);
if (!voice_channel) {
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to create voice channel.");
auto error =
rtp_manager()->GetAudioTransceiver()->internal()->CreateChannel(
voice->name, pc_->call_ptr(), pc_->configuration()->media_config,
pc_->SrtpRequired(), pc_->GetCryptoOptions(), audio_options(),
video_options(), video_bitrate_allocator_factory_.get(),
[&](absl::string_view mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
if (!error.ok()) {
return error;
}
rtp_manager()->GetAudioTransceiver()->internal()->SetChannel(
std::move(voice_channel), [&](const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
}
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
if (video && !video->rejected &&
!rtp_manager()->GetVideoTransceiver()->internal()->channel()) {
std::unique_ptr<cricket::VideoChannel> video_channel =
CreateVideoChannel(video->name);
if (!video_channel) {
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to create video channel.");
auto error =
rtp_manager()->GetVideoTransceiver()->internal()->CreateChannel(
video->name, pc_->call_ptr(), pc_->configuration()->media_config,
pc_->SrtpRequired(), pc_->GetCryptoOptions(),
audio_options(), video_options(),
video_bitrate_allocator_factory_.get(), [&](absl::string_view mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
if (!error.ok()) {
return error;
}
rtp_manager()->GetVideoTransceiver()->internal()->SetChannel(
std::move(video_channel), [&](const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
}
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
@ -4883,39 +4879,6 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
return RTCError::OK();
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
std::unique_ptr<cricket::VoiceChannel>
SdpOfferAnswerHandler::CreateVoiceChannel(const std::string& mid) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateVoiceChannel");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!channel_manager()->media_engine())
return nullptr;
// TODO(bugs.webrtc.org/11992): CreateVoiceChannel internally switches to the
// worker thread. We shouldn't be using the `call_ptr_` hack here but simply
// be on the worker thread and use `call_` (update upstream code).
return channel_manager()->CreateVoiceChannel(
pc_->call_ptr(), pc_->configuration()->media_config, mid,
pc_->SrtpRequired(), pc_->GetCryptoOptions(), audio_options());
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
std::unique_ptr<cricket::VideoChannel>
SdpOfferAnswerHandler::CreateVideoChannel(const std::string& mid) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateVideoChannel");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!channel_manager()->media_engine())
return nullptr;
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to the
// worker thread. We shouldn't be using the `call_ptr_` hack here but simply
// be on the worker thread and use `call_` (update upstream code).
return channel_manager()->CreateVideoChannel(
pc_->call_ptr(), pc_->configuration()->media_config, mid,
pc_->SrtpRequired(), pc_->GetCryptoOptions(), video_options(),
video_bitrate_allocator_factory_.get());
}
bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!context_->network_thread()->Invoke<bool>(RTC_FROM_HERE, [this, &mid] {

5
pc/sdp_offer_answer.h
View File

@ -527,11 +527,6 @@ class SdpOfferAnswerHandler : public SdpStateProvider,
// This method will also delete any existing media channels before creating.
RTCError CreateChannels(const cricket::SessionDescription& desc);
// Helper methods to create media channels.
std::unique_ptr<cricket::VoiceChannel> CreateVoiceChannel(
const std::string& mid);
std::unique_ptr<cricket::VideoChannel> CreateVideoChannel(
const std::string& mid);
bool CreateDataChannel(const std::string& mid);
// Destroys the RTP data channel transport and/or the SCTP data channel