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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_WEBRTCVIDEOENGINE_H_
#define MEDIA_ENGINE_WEBRTCVIDEOENGINE_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_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/mediaengine.h"
#include "media/engine/webrtcvideodecoderfactory.h"
#include "media/engine/webrtcvideoencoderfactory.h"
#include "rtc_base/asyncinvoker.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/networkroute.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class VideoDecoder;
class VideoDecoderFactory;
class VideoEncoder;
class VideoEncoderFactory;
struct MediaConfig;
} // namespace webrtc
namespace rtc {
class Thread;
} // namespace rtc
namespace cricket {
class DecoderFactoryAdapter;
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:
#if defined(USE_BUILTIN_SW_CODECS)
// Internal SW video codecs will be added on top of the external codecs.
WebRtcVideoEngine(
std::unique_ptr<WebRtcVideoEncoderFactory> external_video_encoder_factory,
std::unique_ptr<WebRtcVideoDecoderFactory>
external_video_decoder_factory);
#endif
// 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);
virtual ~WebRtcVideoEngine();
WebRtcVideoChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options);
std::vector<VideoCodec> codecs() const;
RtpCapabilities GetCapabilities() const;
private:
const std::unique_ptr<DecoderFactoryAdapter> decoder_factory_;
const std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory_;
};
class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
public:
WebRtcVideoChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
webrtc::VideoEncoderFactory* encoder_factory,
DecoderFactoryAdapter* decoder_factory);
~WebRtcVideoChannel() override;
// VideoMediaChannel implementation
rtc::DiffServCodePoint PreferredDscp() const override;
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,
const rtc::PacketTime& packet_time) override;
void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
void OnReadyToSend(bool ready) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void SetInterface(NetworkInterface* iface) override;
// Implemented for VideoMediaChannelTest.
bool sending() const { return sending_; }
absl::optional<uint32_t> GetDefaultReceiveStreamSsrc();
StreamParams unsignaled_stream_params() { 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;
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<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;
bool GetChangedRecvParameters(const VideoRecvParameters& params,
ChangedRecvParameters* changed_params) const;
void SetMaxSendBandwidth(int bps);
void ConfigureReceiverRtp(
webrtc::VideoReceiveStream::Config* config,
webrtc::FlexfecReceiveStream::Config* flexfec_config,
const StreamParams& sp) const;
bool ValidateSendSsrcAvailability(const StreamParams& sp) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
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;
// 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();
webrtc::RTCError ValidateRtpParameters(
const webrtc::RtpParameters& parameters);
// 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(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoReceiveStream::Config config,
DecoderFactoryAdapter* 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;
// 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 SetSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
VideoReceiverInfo GetVideoReceiverInfo(bool log_stats);
private:
struct SdpVideoFormatCompare {
bool operator()(const webrtc::SdpVideoFormat& lhs,
const webrtc::SdpVideoFormat& rhs) const {
return std::tie(lhs.name, lhs.parameters) <
std::tie(rhs.name, rhs.parameters);
}
};
typedef std::map<webrtc::SdpVideoFormat,
std::unique_ptr<webrtc::VideoDecoder>,
SdpVideoFormatCompare>
DecoderMap;
void RecreateWebRtcVideoStream();
void MaybeRecreateWebRtcFlexfecStream();
void MaybeAssociateFlexfecWithVideo();
void MaybeDissociateFlexfecFromVideo();
void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs,
DecoderMap* old_codecs);
void ConfigureFlexfecCodec(int flexfec_payload_type);
std::string GetCodecNameFromPayloadType(int payload_type);
absl::optional<uint32_t> GetFirstPrimarySsrc() const;
webrtc::Call* const call_;
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_;
DecoderFactoryAdapter* decoder_factory_;
DecoderMap allocated_decoders_;
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;
static bool NonFlexfecReceiveCodecsHaveChanged(
std::vector<VideoCodecSettings> before,
std::vector<VideoCodecSettings> after);
void FillSenderStats(VideoMediaInfo* info, bool log_stats);
void FillReceiverStats(VideoMediaInfo* info, bool log_stats);
void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
VideoMediaInfo* info);
void FillSendAndReceiveCodecStats(VideoMediaInfo* video_media_info);
rtc::ThreadChecker thread_checker_;
uint32_t rtcp_receiver_report_ssrc_;
bool sending_;
webrtc::Call* const call_;
DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
const MediaConfig::Video video_config_;
rtc::CriticalSection stream_crit_;
// Using primary-ssrc (first ssrc) as key.
std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
RTC_GUARDED_BY(stream_crit_);
std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
RTC_GUARDED_BY(stream_crit_);
std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(stream_crit_);
std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(stream_crit_);
absl::optional<VideoCodecSettings> send_codec_;
absl::optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;
webrtc::VideoEncoderFactory* const encoder_factory_;
DecoderFactoryAdapter* const decoder_factory_;
std::vector<VideoCodecSettings> recv_codecs_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
// See reason for keeping track of the FlexFEC payload type separately in
// comment in WebRtcVideoChannel::ChangedRecvParameters.
int recv_flexfec_payload_type_;
webrtc::BitrateConstraints bitrate_config_;
// TODO(deadbeef): Don't duplicate information between
// send_params/recv_params, rtp_extensions, options, etc.
VideoSendParameters send_params_;
VideoOptions default_send_options_;
VideoRecvParameters recv_params_;
int64_t last_stats_log_ms_;
// 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_;
};
class EncoderStreamFactory
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
EncoderStreamFactory(std::string codec_name,
int max_qp,
int max_framerate,
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 int max_framerate_;
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_WEBRTCVIDEOENGINE_H_