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webrtc/media/engine/webrtc_video_engine.h
Jonas Oreland a3aa9bd75b Make VideoBitrateAllocatorFactory injectable. 
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).

With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.

WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@

Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
2019-04-17 06:17:34 +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/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/async_invoker.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/network_route.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class VideoDecoderFactory;
class VideoEncoderFactory;
struct MediaConfig;
} // namespace webrtc
namespace rtc {
class Thread;
} // namespace rtc
namespace cricket {
class WebRtcVideoChannel;
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);
~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> codecs() const override;
RtpCapabilities GetCapabilities() 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_;
};
class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
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;
bool SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) 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;
bool SetSink(uint32_t ssrc,
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 OnRtcpReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
void OnReadyToSend(bool ready) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void SetInterface(NetworkInterface* iface,
webrtc::MediaTransportInterface* media_transport) 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;
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);
private:
class WebRtcVideoReceiveStream;
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;
int rtx_payload_type;
};
struct ChangedSendParameters {
// These optionals are unset if not changed.
absl::optional<VideoCodecSettings> codec;
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 GetChangedRecvParameters(const VideoRecvParameters& params,
ChangedRecvParameters* changed_params) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void SetMaxSendBandwidth(int bps);
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 rtc::VideoSourceInterface<webrtc::VideoFrame> {
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);
virtual ~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);
// Implements rtc::VideoSourceInterface<webrtc::VideoFrame>.
// WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream
// in |stream_|. This is done to proxy VideoSinkWants from the encoder to
// the worker thread.
void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override;
void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool SetVideoSend(const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
void SetSend(bool send);
const std::vector<uint32_t>& GetSsrcs() const;
VideoSenderInfo GetVideoSenderInfo(bool log_stats);
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
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_);
rtc::ThreadChecker thread_checker_;
rtc::AsyncInvoker invoker_;
rtc::Thread* 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_);
rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
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_);
};
// 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,
webrtc::VideoDecoderFactory* decoder_factory,
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 nack_enabled,
bool remb_enabled,
bool transport_cc_enabled,
webrtc::RtcpMode rtcp_mode);
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);
private:
void RecreateWebRtcVideoStream();
void MaybeRecreateWebRtcFlexfecStream();
void MaybeAssociateFlexfecWithVideo();
void MaybeDissociateFlexfecFromVideo();
void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
void ConfigureFlexfecCodec(int flexfec_payload_type);
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::VideoDecoderFactory* const decoder_factory_;
rtc::CriticalSection 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;
static std::vector<VideoCodecSettings> MapCodecs(
const std::vector<VideoCodec>& codecs);
// Select what video codec will be used for sending, i.e. what codec is used
// for local encoding, based on supported remote codecs. The first remote
// codec that is supported locally will be selected.
absl::optional<VideoCodecSettings> SelectSendVideoCodec(
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_);
rtc::ThreadChecker thread_checker_;
uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
bool sending_ RTC_GUARDED_BY(thread_checker_);
webrtc::Call* const call_ RTC_GUARDED_BY(thread_checker_);
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_);
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_);
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_);
// Buffer for unhandled packets.
std::unique_ptr<UnhandledPacketsBuffer> unknown_ssrc_packet_buffer_
RTC_GUARDED_BY(thread_checker_);
};
class EncoderStreamFactory
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
EncoderStreamFactory(std::string codec_name,
int max_qp,
bool is_screenshare,
bool screenshare_config_explicitly_enabled);
private:
std::vector<webrtc::VideoStream> CreateEncoderStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) override;
const std::string codec_name_;
const int max_qp_;
const bool is_screenshare_;
// Allows a screenshare specific configuration, which enables temporal
// layering and allows simulcast.
const bool screenshare_config_explicitly_enabled_;
};
} // namespace cricket
#endif // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
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