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webrtc/media/engine/webrtc_video_engine.h
Harald Alvestrand d78e30e00b Deprecate cricket::VideoCodec and cricket::AudioCodec 
These are aliases for cricket::Codec.
Also remove internal usage

Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
2024-04-29 16:24:51 +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 <stddef.h>
#include <cstdint>
#include <functional>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/field_trials_view.h"
#include "api/frame_transformer_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/recordable_encoded_frame.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/video_stream_encoder_settings.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/call.h"
#include "call/flexfec_receive_stream.h"
#include "call/rtp_config.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_impl.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/network/sent_packet.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"
#include "video/config/video_encoder_config.h"
namespace webrtc {
class VideoDecoderFactory;
class VideoEncoderFactory;
} // namespace webrtc
namespace cricket {
// 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);
// 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;
std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
override;
std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) override;
std::vector<Codec> send_codecs() const override { return send_codecs(true); }
std::vector<Codec> recv_codecs() const override { return recv_codecs(true); }
std::vector<Codec> send_codecs(bool include_rtx) const override;
std::vector<Codec> 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_;
};
struct VideoCodecSettings {
explicit VideoCodecSettings(const Codec& codec);
// 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);
Codec codec;
webrtc::UlpfecConfig ulpfec;
int flexfec_payload_type; // -1 if absent.
int rtx_payload_type; // -1 if absent.
absl::optional<int> rtx_time;
};
class WebRtcVideoSendChannel : public MediaChannelUtil,
public VideoMediaSendChannelInterface,
public webrtc::EncoderSwitchRequestCallback {
public:
WebRtcVideoSendChannel(
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);
~WebRtcVideoSendChannel() override;
MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
// Type manipulations
VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
// Functions imported from MediaChannelUtil
bool HasNetworkInterface() const override {
return MediaChannelUtil::HasNetworkInterface();
}
void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
MediaChannelUtil::SetExtmapAllowMixed(extmap_allow_mixed);
}
bool ExtmapAllowMixed() const override {
return MediaChannelUtil::ExtmapAllowMixed();
}
// Common functions between sender and receiver
void SetInterface(MediaChannelNetworkInterface* iface) override;
// VideoMediaSendChannelInterface implementation
bool SetSenderParameters(const VideoSenderParameters& params) override;
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
absl::optional<Codec> GetSendCodec() const 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;
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
bool GetStats(VideoMediaSendInfo* info) 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;
// 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 SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {
send_codec_changed_callback_ = std::move(callback);
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
ssrc_list_changed_callback_ = std::move(callback);
}
// Implemented for VideoMediaChannelTest.
bool sending() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
return sending_;
}
// 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,
};
// Implements webrtc::EncoderSwitchRequestCallback.
void RequestEncoderFallback() override;
void RequestEncoderSwitch(const webrtc::SdpVideoFormat& format,
bool allow_default_fallback) 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;
// Information queries to support SetReceiverFeedbackParameters
webrtc::RtcpMode SendCodecRtcpMode() const override {
RTC_DCHECK_RUN_ON(&thread_checker_);
return send_params_.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound;
}
bool SendCodecHasLntf() const override {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
return false;
}
return HasLntf(send_codec()->codec);
}
bool SendCodecHasNack() const override {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
return false;
}
return HasNack(send_codec()->codec);
}
absl::optional<int> SendCodecRtxTime() const override {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
return absl::nullopt;
}
return send_codec()->rtx_time;
}
private:
struct ChangedSenderParameters {
// 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;
};
bool GetChangedSenderParameters(const VideoSenderParameters& params,
ChangedSenderParameters* changed_params) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
bool ApplyChangedParams(const ChangedSenderParameters& changed_params);
bool ValidateSendSsrcAvailability(const StreamParams& sp) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
// 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);
// 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 VideoSenderParameters& send_params);
~WebRtcVideoSendStream();
void SetSenderParameters(const ChangedSenderParameters& 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 Codec& codec);
void SetCodec(const VideoCodecSettings& codec);
void RecreateWebRtcStream();
webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
const Codec& codec) const;
void ReconfigureEncoder(webrtc::SetParametersCallback callback);
// Calls Start or Stop according to whether or not `sending_` is true.
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_;
};
void Construct(webrtc::Call* call, WebRtcVideoEngine* engine);
// 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_);
void FillSenderStats(VideoMediaSendInfo* 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 FillSendCodecStats(VideoMediaSendInfo* video_media_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
// Accessor function for send_codec_. Introduced in order to ensure
// that a receive channel does not touch the send codec directly.
// Can go away once these are different classes.
// TODO(bugs.webrtc.org/13931): Remove this function
absl::optional<VideoCodecSettings>& send_codec() { return send_codec_; }
const absl::optional<VideoCodecSettings>& send_codec() const {
return send_codec_;
}
webrtc::TaskQueueBase* const worker_thread_;
webrtc::ScopedTaskSafety task_safety_;
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_thread_checker_{
webrtc::SequenceChecker::kDetached};
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_);
bool receiving_ RTC_GUARDED_BY(&thread_checker_);
webrtc::Call* const call_;
rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_
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_);
// 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_);
webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_
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::ChangedReceiverParameters.
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.
VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
VideoReceiverParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
int64_t last_send_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
int64_t last_receive_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_);
// RTP parameters that need to be set when creating a video receive stream.
// Only used in Receiver mode - in Both mode, it reads those things from the
// codec.
webrtc::VideoReceiveStreamInterface::Config::Rtp rtp_config_;
// Callback invoked whenever the send codec changes.
// TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
absl::AnyInvocable<void()> send_codec_changed_callback_;
// Callback invoked whenever the list of SSRCs changes.
absl::AnyInvocable<void(const std::set<uint32_t>&)>
ssrc_list_changed_callback_;
};
class WebRtcVideoReceiveChannel : public MediaChannelUtil,
public VideoMediaReceiveChannelInterface {
public:
WebRtcVideoReceiveChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoDecoderFactory* decoder_factory);
~WebRtcVideoReceiveChannel() override;
public:
MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
return this;
}
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
// Common functions between sender and receiver
void SetInterface(MediaChannelNetworkInterface* iface) override;
// VideoMediaReceiveChannelInterface implementation
bool SetReceiverParameters(const VideoReceiverParameters& params) override;
webrtc::RtpParameters GetRtpReceiverParameters(uint32_t ssrc) const override;
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
void SetReceive(bool receive) override;
bool AddRecvStream(const StreamParams& sp) override;
bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
// Invokes private AddRecvStream variant function
return AddRecvStream(sp, true);
}
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
absl::optional<uint32_t> GetUnsignaledSsrc() const 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;
bool GetStats(VideoMediaReceiveInfo* info) override;
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const override;
// Choose one of the available SSRCs (or default if none) as the current
// receiver report SSRC.
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) 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;
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 SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
absl::optional<int> rtx_time) override;
private:
class WebRtcVideoReceiveStream;
struct ChangedReceiverParameters {
// 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;
};
// Finds VideoReceiveStreamInterface corresponding to ssrc. Aware of
// unsignalled ssrc handling.
WebRtcVideoReceiveStream* FindReceiveStream(uint32_t ssrc)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void ProcessReceivedPacket(webrtc::RtpPacketReceived packet)
RTC_RUN_ON(thread_checker_);
// Expected to be invoked once per packet that belongs to this channel that
// can not be demuxed.
// Returns true if a new default stream has been created.
bool MaybeCreateDefaultReceiveStream(
const webrtc::RtpPacketReceived& parsed_packet)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void ReCreateDefaultReceiveStream(uint32_t ssrc,
absl::optional<uint32_t> rtx_ssrc);
// Add a receive stream. Used for testing.
bool AddRecvStream(const StreamParams& sp, bool default_stream);
void ConfigureReceiverRtp(
webrtc::VideoReceiveStreamInterface::Config* config,
webrtc::FlexfecReceiveStream::Config* flexfec_config,
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_);
// 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 receiver part, contains configs etc. that are needed to
// reconstruct the underlying VideoReceiveStreamInterface.
class WebRtcVideoReceiveStream
: public rtc::VideoSinkInterface<webrtc::VideoFrame> {
public:
WebRtcVideoReceiveStream(
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, nor header extensions, 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,
webrtc::RtcpMode rtcp_mode,
absl::optional<int> rtx_time);
void SetReceiverParameters(const ChangedReceiverParameters& 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);
void UpdateRtxSsrc(uint32_t ssrc);
void StartReceiveStream();
void StopReceiveStream();
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();
// 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);
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_);
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_);
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
bool receiving_ RTC_GUARDED_BY(&thread_checker_);
};
bool GetChangedReceiverParameters(const VideoReceiverParameters& params,
ChangedReceiverParameters* changed_params)
const RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
RTC_GUARDED_BY(thread_checker_);
void FillReceiverStats(VideoMediaReceiveInfo* info, bool log_stats)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void FillReceiveCodecStats(VideoMediaReceiveInfo* video_media_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
StreamParams unsignaled_stream_params() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return unsignaled_stream_params_;
}
// Variables.
webrtc::TaskQueueBase* const worker_thread_;
webrtc::ScopedTaskSafety task_safety_;
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_thread_checker_{
webrtc::SequenceChecker::kDetached};
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
bool receiving_ RTC_GUARDED_BY(&thread_checker_);
webrtc::Call* const call_;
rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_
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_);
// 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::VideoDecoderFactory* const decoder_factory_
RTC_GUARDED_BY(thread_checker_);
std::vector<VideoCodecSettings> recv_codecs_ RTC_GUARDED_BY(thread_checker_);
webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_
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::ChangedReceiverParameters.
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.
VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
VideoReceiverParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
int64_t last_receive_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_);
// RTP parameters that need to be set when creating a video receive stream.
// Only used in Receiver mode - in Both mode, it reads those things from the
// codec.
webrtc::VideoReceiveStreamInterface::Config::Rtp rtp_config_;
// Callback invoked whenever the send codec changes.
// TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
absl::AnyInvocable<void()> send_codec_changed_callback_;
// Callback invoked whenever the list of SSRCs changes.
absl::AnyInvocable<void(const std::set<uint32_t>&)>
ssrc_list_changed_callback_;
const int receive_buffer_size_;
};
// Keeping the old name "WebRtcVideoChannel" around because some external
// customers are using cricket::WebRtcVideoChannel::AdaptReason
// TODO(bugs.webrtc.org/15216): Move this enum to an interface class and
// delete this workaround.
class WebRtcVideoChannel : public WebRtcVideoSendChannel {
public:
// Make all the values of AdaptReason available as
// WebRtcVideoChannel::ADAPT_xxx.
using WebRtcVideoSendChannel::AdaptReason;
};
} // namespace cricket
#endif // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
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