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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/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/task_utils/pending_task_safety_flag.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) = 0;
virtual ~UnsignalledSsrcHandler() = default;
};
// TODO(pbos): Remove, use external handlers only.
class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
public:
DefaultUnsignalledSsrcHandler();
Action OnUnsignalledSsrc(WebRtcVideoChannel* channel, uint32_t 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::WebRtcKeyValueConfig& 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;
std::vector<VideoCodec> recv_codecs() 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::WebRtcKeyValueConfig& 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) 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(const std::string& 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;
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;
// TODO(bugs.webrtc.org/11341) : Remove this version of RequestEncoderSwitch.
void RequestEncoderSwitch(
const EncoderSwitchRequestCallback::Config& conf) override;
void RequestEncoderSwitch(const webrtc::SdpVideoFormat& format) override;
void SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
override;
void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
void GenerateKeyFrame(uint32_t ssrc) 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 VideoReceiveStream 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
// VideoReceiveStream 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::VideoReceiveStream::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);
// 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::RtpParameters GetRtpParameters() const;
void SetFrameEncryptor(
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor);
bool SetVideoSend(const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
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);
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();
// 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_);
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 VideoReceiveStream.
class WebRtcVideoReceiveStream
: public rtc::VideoSinkInterface<webrtc::VideoFrame> {
public:
WebRtcVideoReceiveStream(
WebRtcVideoChannel* channel,
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoReceiveStream::Config config,
bool default_stream,
const std::vector<VideoCodecSettings>& recv_codecs,
const webrtc::FlexfecReceiveStream::Config& flexfec_config);
~WebRtcVideoReceiveStream();
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;
void SetLocalSsrc(uint32_t local_ssrc);
// 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);
private:
void RecreateWebRtcVideoStream();
// 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 ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
std::string GetCodecNameFromPayloadType(int payload_type);
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::VideoReceiveStream* stream_;
const bool default_stream_;
webrtc::VideoReceiveStream::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_;
webrtc::SequenceChecker network_thread_checker_;
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_);
absl::optional<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_);
bool allow_codec_switching_ = false;
absl::optional<EncoderSwitchRequestCallback::Config>
requested_encoder_switch_;
};
class EncoderStreamFactory
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
EncoderStreamFactory(std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode)
: EncoderStreamFactory(codec_name,
max_qp,
is_screenshare,
conference_mode,
nullptr) {}
EncoderStreamFactory(std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode,
const webrtc::WebRtcKeyValueConfig* trials);
private:
std::vector<webrtc::VideoStream> CreateEncoderStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) override;
std::vector<webrtc::VideoStream> CreateDefaultVideoStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const;
std::vector<webrtc::VideoStream>
CreateSimulcastOrConferenceModeScreenshareStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const;
const std::string codec_name_;
const int max_qp_;
const bool is_screenshare_;
// Allows a screenshare specific configuration, which enables temporal
// layering and various settings.
const bool conference_mode_;
const webrtc::FieldTrialBasedConfig fallback_trials_;
const webrtc::WebRtcKeyValueConfig& trials_;
};
} // namespace cricket
#endif // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_