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webrtc/media/engine/webrtc_video_engine.h
Florent Castelli acabb3641b pc: Add asynchronous RtpSender::SetParameters() call 
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.

Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
2022-11-15 15:31:40 +00:00

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/*
* Copyright (c) 2014 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 MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
#define MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/call.h"
#include "call/flexfec_receive_stream.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "media/base/media_engine.h"
#include "media/engine/unhandled_packets_buffer.h"
#include "rtc_base/network_route.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class VideoDecoderFactory;
class VideoEncoderFactory;
} // namespace webrtc
namespace cricket {
class WebRtcVideoChannel;
// Public for testing.
// Inputs StreamStats for all types of substreams (kMedia, kRtx, kFlexfec) and
// merges any non-kMedia substream stats object into its referenced kMedia-type
// substream. The resulting substreams are all kMedia. This means, for example,
// that packet and byte counters of RTX and FlexFEC streams are accounted for in
// the relevant RTP media stream's stats. This makes the resulting StreamStats
// objects ready to be turned into "outbound-rtp" stats objects for GetStats()
// which does not create separate stream stats objects for complementary
// streams.
std::map<uint32_t, webrtc::VideoSendStream::StreamStats>
MergeInfoAboutOutboundRtpSubstreamsForTesting(
const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>& substreams);
class UnsignalledSsrcHandler {
public:
enum Action {
kDropPacket,
kDeliverPacket,
};
virtual Action OnUnsignalledSsrc(WebRtcVideoChannel* channel,
uint32_t ssrc,
absl::optional<uint32_t> rtx_ssrc) = 0;
virtual ~UnsignalledSsrcHandler() = default;
};
// TODO(pbos): Remove, use external handlers only.
class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
public:
DefaultUnsignalledSsrcHandler();
Action OnUnsignalledSsrc(WebRtcVideoChannel* channel,
uint32_t ssrc,
absl::optional<uint32_t> rtx_ssrc) override;
rtc::VideoSinkInterface<webrtc::VideoFrame>* GetDefaultSink() const;
void SetDefaultSink(WebRtcVideoChannel* channel,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
virtual ~DefaultUnsignalledSsrcHandler() = default;
private:
rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_;
};
// WebRtcVideoEngine is used for the new native WebRTC Video API (webrtc:1667).
class WebRtcVideoEngine : public VideoEngineInterface {
public:
// These video codec factories represents all video codecs, i.e. both software
// and external hardware codecs.
WebRtcVideoEngine(
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
const webrtc::FieldTrialsView& trials);
~WebRtcVideoEngine() override;
VideoMediaChannel* CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
override;
std::vector<VideoCodec> send_codecs() const override {
return send_codecs(true);
}
std::vector<VideoCodec> recv_codecs() const override {
return recv_codecs(true);
}
std::vector<VideoCodec> send_codecs(bool include_rtx) const override;
std::vector<VideoCodec> recv_codecs(bool include_rtx) const override;
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
const override;
private:
const std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory_;
const std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory_;
const std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
bitrate_allocator_factory_;
const webrtc::FieldTrialsView& trials_;
};
class WebRtcVideoChannel : public VideoMediaChannel,
public webrtc::Transport,
public webrtc::EncoderSwitchRequestCallback {
public:
WebRtcVideoChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoEncoderFactory* encoder_factory,
webrtc::VideoDecoderFactory* decoder_factory,
webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory);
~WebRtcVideoChannel() override;
// VideoMediaChannel implementation
bool SetSendParameters(const VideoSendParameters& params) override;
bool SetRecvParameters(const VideoRecvParameters& params) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
bool GetSendCodec(VideoCodec* send_codec) override;
bool SetSend(bool send) override;
bool SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32_t ssrc) override;
bool AddRecvStream(const StreamParams& sp) override;
bool AddRecvStream(const StreamParams& sp, bool default_stream);
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
void SetDefaultSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
bool GetStats(VideoMediaInfo* info) override;
void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnPacketSent(const rtc::SentPacket& sent_packet) override;
void OnReadyToSend(bool ready) override;
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override;
void SetInterface(NetworkInterface* iface) override;
// E2E Encrypted Video Frame API
// Set a frame decryptor to a particular ssrc that will intercept all
// incoming video frames and attempt to decrypt them before forwarding the
// result.
void SetFrameDecryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
// Set a frame encryptor to a particular ssrc that will intercept all
// outgoing video frames and attempt to encrypt them and forward the result
// to the packetizer.
void SetFrameEncryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
frame_encryptor) override;
// note: The encoder_selector object must remain valid for the lifetime of the
// MediaChannel, unless replaced.
void SetEncoderSelector(uint32_t ssrc,
webrtc::VideoEncoderFactory::EncoderSelectorInterface*
encoder_selector) override;
void SetVideoCodecSwitchingEnabled(bool enabled) override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const override;
// Implemented for VideoMediaChannelTest.
bool sending() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
return sending_;
}
absl::optional<uint32_t> GetDefaultReceiveStreamSsrc();
StreamParams unsignaled_stream_params() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return unsignaled_stream_params_;
}
// AdaptReason is used for expressing why a WebRtcVideoSendStream request
// a lower input frame size than the currently configured camera input frame
// size. There can be more than one reason OR:ed together.
enum AdaptReason {
ADAPTREASON_NONE = 0,
ADAPTREASON_CPU = 1,
ADAPTREASON_BANDWIDTH = 2,
};
static constexpr int kDefaultQpMax = 56;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
// Take the buffered packets for `ssrcs` and feed them into DeliverPacket.
// This method does nothing unless unknown_ssrc_packet_buffer_ is configured.
void BackfillBufferedPackets(rtc::ArrayView<const uint32_t> ssrcs);
// Implements webrtc::EncoderSwitchRequestCallback.
void RequestEncoderFallback() override;
void RequestEncoderSwitch(const webrtc::SdpVideoFormat& format,
bool allow_default_fallback) override;
void SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
override;
void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
void RequestRecvKeyFrame(uint32_t ssrc) override;
void GenerateSendKeyFrame(uint32_t ssrc,
const std::vector<std::string>& rids) override;
void SetEncoderToPacketizerFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
void SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
private:
class WebRtcVideoReceiveStream;
// Finds VideoReceiveStreamInterface corresponding to ssrc. Aware of
// unsignalled ssrc handling.
WebRtcVideoReceiveStream* FindReceiveStream(uint32_t ssrc)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
struct VideoCodecSettings {
VideoCodecSettings();
// Checks if all members of |*this| are equal to the corresponding members
// of `other`.
bool operator==(const VideoCodecSettings& other) const;
bool operator!=(const VideoCodecSettings& other) const;
// Checks if all members of `a`, except `flexfec_payload_type`, are equal
// to the corresponding members of `b`.
static bool EqualsDisregardingFlexfec(const VideoCodecSettings& a,
const VideoCodecSettings& b);
VideoCodec codec;
webrtc::UlpfecConfig ulpfec;
int flexfec_payload_type; // -1 if absent.
int rtx_payload_type; // -1 if absent.
int rtx_time; // -1 if absent.
};
struct ChangedSendParameters {
// These optionals are unset if not changed.
absl::optional<VideoCodecSettings> send_codec;
absl::optional<std::vector<VideoCodecSettings>> negotiated_codecs;
absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
absl::optional<std::string> mid;
absl::optional<bool> extmap_allow_mixed;
absl::optional<int> max_bandwidth_bps;
absl::optional<bool> conference_mode;
absl::optional<webrtc::RtcpMode> rtcp_mode;
};
struct ChangedRecvParameters {
// These optionals are unset if not changed.
absl::optional<std::vector<VideoCodecSettings>> codec_settings;
absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
// Keep track of the FlexFEC payload type separately from `codec_settings`.
// This allows us to recreate the FlexfecReceiveStream separately from the
// VideoReceiveStreamInterface when the FlexFEC payload type is changed.
absl::optional<int> flexfec_payload_type;
};
bool GetChangedSendParameters(const VideoSendParameters& params,
ChangedSendParameters* changed_params) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
bool ApplyChangedParams(const ChangedSendParameters& changed_params);
bool GetChangedRecvParameters(const VideoRecvParameters& params,
ChangedRecvParameters* changed_params) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void ConfigureReceiverRtp(
webrtc::VideoReceiveStreamInterface::Config* config,
webrtc::FlexfecReceiveStream::Config* flexfec_config,
const StreamParams& sp) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
bool ValidateSendSsrcAvailability(const StreamParams& sp) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
static std::string CodecSettingsVectorToString(
const std::vector<VideoCodecSettings>& codecs);
// Populates `rtx_associated_payload_types`, `raw_payload_types` and
// `decoders` based on codec settings provided by `recv_codecs`.
// `recv_codecs` must be non-empty and all other parameters must be empty.
static void ExtractCodecInformation(
rtc::ArrayView<const VideoCodecSettings> recv_codecs,
std::map<int, int>& rtx_associated_payload_types,
std::set<int>& raw_payload_types,
std::vector<webrtc::VideoReceiveStreamInterface::Decoder>& decoders);
// Called when the local ssrc changes. Sets `rtcp_receiver_report_ssrc_` and
// updates the receive streams.
void SetReceiverReportSsrc(uint32_t ssrc) RTC_RUN_ON(&thread_checker_);
// Wrapper for the sender part.
class WebRtcVideoSendStream {
public:
WebRtcVideoSendStream(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
bool enable_cpu_overuse_detection,
int max_bitrate_bps,
const absl::optional<VideoCodecSettings>& codec_settings,
const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
const VideoSendParameters& send_params);
~WebRtcVideoSendStream();
void SetSendParameters(const ChangedSendParameters& send_params);
webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback);
webrtc::RtpParameters GetRtpParameters() const;
void SetFrameEncryptor(
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor);
bool SetVideoSend(const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
// note: The encoder_selector object must remain valid for the lifetime of
// the MediaChannel, unless replaced.
void SetEncoderSelector(
webrtc::VideoEncoderFactory::EncoderSelectorInterface*
encoder_selector);
void SetSend(bool send);
const std::vector<uint32_t>& GetSsrcs() const;
// Returns per ssrc VideoSenderInfos. Useful for simulcast scenario.
std::vector<VideoSenderInfo> GetPerLayerVideoSenderInfos(bool log_stats);
// Aggregates per ssrc VideoSenderInfos to single VideoSenderInfo for
// legacy reasons. Used in old GetStats API and track stats.
VideoSenderInfo GetAggregatedVideoSenderInfo(
const std::vector<VideoSenderInfo>& infos) const;
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
void SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer);
void GenerateKeyFrame(const std::vector<std::string>& rids);
private:
// Parameters needed to reconstruct the underlying stream.
// webrtc::VideoSendStream doesn't support setting a lot of options on the
// fly, so when those need to be changed we tear down and reconstruct with
// similar parameters depending on which options changed etc.
struct VideoSendStreamParameters {
VideoSendStreamParameters(
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
int max_bitrate_bps,
const absl::optional<VideoCodecSettings>& codec_settings);
webrtc::VideoSendStream::Config config;
VideoOptions options;
int max_bitrate_bps;
bool conference_mode;
absl::optional<VideoCodecSettings> codec_settings;
// Sent resolutions + bitrates etc. by the underlying VideoSendStream,
// typically changes when setting a new resolution or reconfiguring
// bitrates.
webrtc::VideoEncoderConfig encoder_config;
};
rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
ConfigureVideoEncoderSettings(const VideoCodec& codec);
void SetCodec(const VideoCodecSettings& codec);
void RecreateWebRtcStream();
webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
const VideoCodec& codec) const;
void ReconfigureEncoder(webrtc::SetParametersCallback callback);
// Calls Start or Stop according to whether or not `sending_` is true,
// and whether or not the encoding in `rtp_parameters_` is active.
void UpdateSendState();
webrtc::DegradationPreference GetDegradationPreference() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
webrtc::TaskQueueBase* const worker_thread_;
const std::vector<uint32_t> ssrcs_ RTC_GUARDED_BY(&thread_checker_);
const std::vector<SsrcGroup> ssrc_groups_ RTC_GUARDED_BY(&thread_checker_);
webrtc::Call* const call_;
const bool enable_cpu_overuse_detection_;
rtc::VideoSourceInterface<webrtc::VideoFrame>* source_
RTC_GUARDED_BY(&thread_checker_);
webrtc::VideoSendStream* stream_ RTC_GUARDED_BY(&thread_checker_);
// Contains settings that are the same for all streams in the MediaChannel,
// such as codecs, header extensions, and the global bitrate limit for the
// entire channel.
VideoSendStreamParameters parameters_ RTC_GUARDED_BY(&thread_checker_);
// Contains settings that are unique for each stream, such as max_bitrate.
// Does *not* contain codecs, however.
// TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
// TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
// one stream per MediaChannel.
webrtc::RtpParameters rtp_parameters_ RTC_GUARDED_BY(&thread_checker_);
bool sending_ RTC_GUARDED_BY(&thread_checker_);
// TODO(asapersson): investigate why setting
// DegrationPreferences::MAINTAIN_RESOLUTION isn't sufficient to disable
// downscaling everywhere in the pipeline.
const bool disable_automatic_resize_;
};
// Wrapper for the receiver part, contains configs etc. that are needed to
// reconstruct the underlying VideoReceiveStreamInterface.
class WebRtcVideoReceiveStream
: public rtc::VideoSinkInterface<webrtc::VideoFrame> {
public:
WebRtcVideoReceiveStream(
WebRtcVideoChannel* channel,
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoReceiveStreamInterface::Config config,
bool default_stream,
const std::vector<VideoCodecSettings>& recv_codecs,
const webrtc::FlexfecReceiveStream::Config& flexfec_config);
~WebRtcVideoReceiveStream();
webrtc::VideoReceiveStreamInterface& stream();
// Return value may be nullptr.
webrtc::FlexfecReceiveStream* flexfec_stream();
const std::vector<uint32_t>& GetSsrcs() const;
std::vector<webrtc::RtpSource> GetSources();
// Does not return codecs, they are filled by the owning WebRtcVideoChannel.
webrtc::RtpParameters GetRtpParameters() const;
// TODO(deadbeef): Move these feedback parameters into the recv parameters.
void SetFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
bool transport_cc_enabled,
webrtc::RtcpMode rtcp_mode,
int rtx_time);
void SetRecvParameters(const ChangedRecvParameters& recv_params);
void OnFrame(const webrtc::VideoFrame& frame) override;
bool IsDefaultStream() const;
void SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor);
bool SetBaseMinimumPlayoutDelayMs(int delay_ms);
int GetBaseMinimumPlayoutDelayMs() const;
void SetSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
VideoReceiverInfo GetVideoReceiverInfo(bool log_stats);
void SetRecordableEncodedFrameCallback(
std::function<void(const webrtc::RecordableEncodedFrame&)> callback);
void ClearRecordableEncodedFrameCallback();
void GenerateKeyFrame();
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer);
void SetLocalSsrc(uint32_t local_ssrc);
private:
// Attempts to reconfigure an already existing `flexfec_stream_`, create
// one if the configuration is now complete or remove a flexfec stream
// when disabled.
void SetFlexFecPayload(int payload_type);
void RecreateReceiveStream();
void CreateReceiveStream();
void StartReceiveStream();
// Applies a new receive codecs configration to `config_`. Returns true
// if the internal stream needs to be reconstructed, or false if no changes
// were applied.
bool ReconfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
WebRtcVideoChannel* const channel_;
webrtc::Call* const call_;
const StreamParams stream_params_;
// Both `stream_` and `flexfec_stream_` are managed by `this`. They are
// destroyed by calling call_->DestroyVideoReceiveStream and
// call_->DestroyFlexfecReceiveStream, respectively.
webrtc::VideoReceiveStreamInterface* stream_;
const bool default_stream_;
webrtc::VideoReceiveStreamInterface::Config config_;
webrtc::FlexfecReceiveStream::Config flexfec_config_;
webrtc::FlexfecReceiveStream* flexfec_stream_;
webrtc::Mutex sink_lock_;
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink_
RTC_GUARDED_BY(sink_lock_);
// Expands remote RTP timestamps to int64_t to be able to estimate how long
// the stream has been running.
rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
RTC_GUARDED_BY(sink_lock_);
int64_t first_frame_timestamp_ RTC_GUARDED_BY(sink_lock_);
// Start NTP time is estimated as current remote NTP time (estimated from
// RTCP) minus the elapsed time, as soon as remote NTP time is available.
int64_t estimated_remote_start_ntp_time_ms_ RTC_GUARDED_BY(sink_lock_);
};
void Construct(webrtc::Call* call, WebRtcVideoEngine* engine);
bool SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
// Generate the list of codec parameters to pass down based on the negotiated
// "codecs". Note that VideoCodecSettings correspond to concrete codecs like
// VP8, VP9, H264 while VideoCodecs correspond also to "virtual" codecs like
// RTX, ULPFEC, FLEXFEC.
static std::vector<VideoCodecSettings> MapCodecs(
const std::vector<VideoCodec>& codecs);
// Get all codecs that are compatible with the receiver.
std::vector<VideoCodecSettings> SelectSendVideoCodecs(
const std::vector<VideoCodecSettings>& remote_mapped_codecs) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
static bool NonFlexfecReceiveCodecsHaveChanged(
std::vector<VideoCodecSettings> before,
std::vector<VideoCodecSettings> after);
void FillSenderStats(VideoMediaInfo* info, bool log_stats)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void FillReceiverStats(VideoMediaInfo* info, bool log_stats)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
VideoMediaInfo* info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void FillSendAndReceiveCodecStats(VideoMediaInfo* video_media_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
webrtc::TaskQueueBase* const worker_thread_;
webrtc::ScopedTaskSafety task_safety_;
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_thread_checker_;
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
bool sending_ RTC_GUARDED_BY(thread_checker_);
webrtc::Call* const call_;
DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_
RTC_GUARDED_BY(thread_checker_);
UnsignalledSsrcHandler* const unsignalled_ssrc_handler_
RTC_GUARDED_BY(thread_checker_);
// Delay for unsignaled streams, which may be set before the stream exists.
int default_recv_base_minimum_delay_ms_ RTC_GUARDED_BY(thread_checker_) = 0;
const MediaConfig::Video video_config_ RTC_GUARDED_BY(thread_checker_);
// Using primary-ssrc (first ssrc) as key.
std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
RTC_GUARDED_BY(thread_checker_);
std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
RTC_GUARDED_BY(thread_checker_);
// When the channel and demuxer get reconfigured, there is a window of time
// where we have to be prepared for packets arriving based on the old demuxer
// criteria because the streams live on the worker thread and the demuxer
// lives on the network thread. Because packets are posted from the network
// thread to the worker thread, they can still be in-flight when streams are
// reconfgured. This can happen when `demuxer_criteria_id_` and
// `demuxer_criteria_completed_id_` don't match. During this time, we do not
// want to create unsignalled receive streams and should instead drop the
// packets. E.g:
// * If RemoveRecvStream(old_ssrc) was recently called, there may be packets
// in-flight for that ssrc. This happens when a receiver becomes inactive.
// * If we go from one to many m= sections, the demuxer may change from
// forwarding all packets to only forwarding the configured ssrcs, so there
// is a risk of receiving ssrcs for other, recently added m= sections.
uint32_t demuxer_criteria_id_ RTC_GUARDED_BY(thread_checker_) = 0;
uint32_t demuxer_criteria_completed_id_ RTC_GUARDED_BY(thread_checker_) = 0;
absl::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
RTC_GUARDED_BY(thread_checker_);
std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(thread_checker_);
absl::optional<VideoCodecSettings> send_codec_
RTC_GUARDED_BY(thread_checker_);
std::vector<VideoCodecSettings> negotiated_codecs_
RTC_GUARDED_BY(thread_checker_);
std::vector<webrtc::RtpExtension> send_rtp_extensions_
RTC_GUARDED_BY(thread_checker_);
webrtc::VideoEncoderFactory* const encoder_factory_
RTC_GUARDED_BY(thread_checker_);
webrtc::VideoDecoderFactory* const decoder_factory_
RTC_GUARDED_BY(thread_checker_);
webrtc::VideoBitrateAllocatorFactory* const bitrate_allocator_factory_
RTC_GUARDED_BY(thread_checker_);
std::vector<VideoCodecSettings> recv_codecs_ RTC_GUARDED_BY(thread_checker_);
std::vector<webrtc::RtpExtension> recv_rtp_extensions_
RTC_GUARDED_BY(thread_checker_);
// See reason for keeping track of the FlexFEC payload type separately in
// comment in WebRtcVideoChannel::ChangedRecvParameters.
int recv_flexfec_payload_type_ RTC_GUARDED_BY(thread_checker_);
webrtc::BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
// TODO(deadbeef): Don't duplicate information between
// send_params/recv_params, rtp_extensions, options, etc.
VideoSendParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
VideoRecvParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
int64_t last_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
const bool discard_unknown_ssrc_packets_ RTC_GUARDED_BY(thread_checker_);
// This is a stream param that comes from the remote description, but wasn't
// signaled with any a=ssrc lines. It holds information that was signaled
// before the unsignaled receive stream is created when the first packet is
// received.
StreamParams unsignaled_stream_params_ RTC_GUARDED_BY(thread_checker_);
// Per peer connection crypto options that last for the lifetime of the peer
// connection.
const webrtc::CryptoOptions crypto_options_ RTC_GUARDED_BY(thread_checker_);
// Optional frame transformer set on unsignaled streams.
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
unsignaled_frame_transformer_ RTC_GUARDED_BY(thread_checker_);
// Buffer for unhandled packets.
std::unique_ptr<UnhandledPacketsBuffer> unknown_ssrc_packet_buffer_
RTC_GUARDED_BY(thread_checker_);
// TODO(bugs.webrtc.org/11341): Remove this and relevant PC API. Presence
// of multiple negotiated codecs allows generic encoder fallback on failures.
// Presence of EncoderSelector allows switching to specific encoders.
bool allow_codec_switching_ = false;
};
} // namespace cricket
#endif // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
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