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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 WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
#define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "webrtc/base/asyncinvoker.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/networkroute.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/call.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/media/engine/webrtcvideochannelfactory.h"
#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
#include "webrtc/transport.h"
#include "webrtc/video_frame.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
class VideoDecoder;
class VideoEncoder;
struct MediaConfig;
}
namespace rtc {
class Thread;
} // namespace rtc
namespace cricket {
class VideoCapturer;
class VideoFrame;
class VideoProcessor;
class VideoRenderer;
class VoiceMediaChannel;
class WebRtcDecoderObserver;
class WebRtcEncoderObserver;
class WebRtcLocalStreamInfo;
class WebRtcRenderAdapter;
class WebRtcVideoChannelRecvInfo;
class WebRtcVideoChannelSendInfo;
class WebRtcVoiceEngine;
class WebRtcVoiceMediaChannel;
struct CapturedFrame;
struct Device;
// Exposed here for unittests.
std::vector<VideoCodec> DefaultVideoCodecList();
class UnsignalledSsrcHandler {
public:
enum Action {
kDropPacket,
kDeliverPacket,
};
virtual Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
uint32_t ssrc) = 0;
virtual ~UnsignalledSsrcHandler() = default;
};
// TODO(pbos): Remove, use external handlers only.
class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
public:
DefaultUnsignalledSsrcHandler();
Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
uint32_t ssrc) override;
rtc::VideoSinkInterface<VideoFrame>* GetDefaultSink() const;
void SetDefaultSink(VideoMediaChannel* channel,
rtc::VideoSinkInterface<VideoFrame>* sink);
virtual ~DefaultUnsignalledSsrcHandler() = default;
private:
uint32_t default_recv_ssrc_;
rtc::VideoSinkInterface<VideoFrame>* default_sink_;
};
// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
class WebRtcVideoEngine2 {
public:
WebRtcVideoEngine2();
virtual ~WebRtcVideoEngine2();
// Basic video engine implementation.
void Init();
WebRtcVideoChannel2* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options);
const std::vector<VideoCodec>& codecs() const;
RtpCapabilities GetCapabilities() const;
// Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
// not take the ownership of |decoder_factory|. The caller needs to make sure
// that |decoder_factory| outlives the video engine.
void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory);
// Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
// not take the ownership of |encoder_factory|. The caller needs to make sure
// that |encoder_factory| outlives the video engine.
virtual void SetExternalEncoderFactory(
WebRtcVideoEncoderFactory* encoder_factory);
private:
std::vector<VideoCodec> GetSupportedCodecs() const;
std::vector<VideoCodec> video_codecs_;
bool initialized_;
WebRtcVideoDecoderFactory* external_decoder_factory_;
WebRtcVideoEncoderFactory* external_encoder_factory_;
std::unique_ptr<WebRtcVideoEncoderFactory> simulcast_encoder_factory_;
};
class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
public:
WebRtcVideoChannel2(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const std::vector<VideoCodec>& recv_codecs,
WebRtcVideoEncoderFactory* external_encoder_factory,
WebRtcVideoDecoderFactory* external_decoder_factory);
~WebRtcVideoChannel2() 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;
bool 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,
bool enable,
const VideoOptions* options,
rtc::VideoSourceInterface<cricket::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<VideoFrame>* sink) 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_; }
// 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,
};
private:
class WebRtcVideoReceiveStream;
struct VideoCodecSettings {
VideoCodecSettings();
bool operator==(const VideoCodecSettings& other) const;
bool operator!=(const VideoCodecSettings& other) const;
VideoCodec codec;
webrtc::FecConfig fec;
int rtx_payload_type;
};
struct ChangedSendParameters {
// These optionals are unset if not changed.
rtc::Optional<VideoCodecSettings> codec;
rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
rtc::Optional<int> max_bandwidth_bps;
rtc::Optional<bool> conference_mode;
rtc::Optional<webrtc::RtcpMode> rtcp_mode;
};
struct ChangedRecvParameters {
// These optionals are unset if not changed.
rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
};
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,
const StreamParams& sp) const;
bool CodecIsExternallySupported(const std::string& name) const;
bool ValidateSendSsrcAvailability(const StreamParams& sp) const
EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
static std::string CodecSettingsVectorToString(
const std::vector<VideoCodecSettings>& codecs);
// Wrapper for the sender part, this is where the source is connected and
// frames are then converted from cricket frames to webrtc frames.
class WebRtcVideoSendStream
: public rtc::VideoSinkInterface<cricket::VideoFrame>,
public webrtc::LoadObserver {
public:
WebRtcVideoSendStream(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
WebRtcVideoEncoderFactory* external_encoder_factory,
bool enable_cpu_overuse_detection,
int max_bitrate_bps,
const rtc::Optional<VideoCodecSettings>& codec_settings,
const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
const VideoSendParameters& send_params);
virtual ~WebRtcVideoSendStream();
void SetSendParameters(const ChangedSendParameters& send_params);
bool SetRtpParameters(const webrtc::RtpParameters& parameters);
webrtc::RtpParameters GetRtpParameters() const;
void OnFrame(const cricket::VideoFrame& frame) override;
bool SetVideoSend(bool mute,
const VideoOptions* options,
rtc::VideoSourceInterface<cricket::VideoFrame>* source);
void DisconnectSource();
void SetSend(bool send);
// Implements webrtc::LoadObserver.
void OnLoadUpdate(Load load) override;
const std::vector<uint32_t>& GetSsrcs() const;
VideoSenderInfo GetVideoSenderInfo(bool log_stats);
void FillBandwidthEstimationInfo(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 rtc::Optional<VideoCodecSettings>& codec_settings);
webrtc::VideoSendStream::Config config;
VideoOptions options;
int max_bitrate_bps;
bool conference_mode;
rtc::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;
};
struct AllocatedEncoder {
AllocatedEncoder(webrtc::VideoEncoder* encoder,
webrtc::VideoCodecType type,
bool external);
webrtc::VideoEncoder* encoder;
webrtc::VideoEncoder* external_encoder;
webrtc::VideoCodecType type;
bool external;
};
struct VideoFrameInfo {
// Initial encoder configuration (QCIF, 176x144) frame (to ensure that
// hardware encoders can be initialized). This gives us low memory usage
// but also makes it so configuration errors are discovered at the time we
// apply the settings rather than when we get the first frame (waiting for
// the first frame to know that you gave a bad codec parameter could make
// debugging hard).
// TODO(pbos): Consider setting up encoders lazily.
VideoFrameInfo()
: width(176),
height(144),
rotation(webrtc::kVideoRotation_0),
is_texture(false) {}
int width;
int height;
webrtc::VideoRotation rotation;
bool is_texture;
};
union VideoEncoderSettings {
webrtc::VideoCodecH264 h264;
webrtc::VideoCodecVP8 vp8;
webrtc::VideoCodecVP9 vp9;
};
static std::vector<webrtc::VideoStream> CreateVideoStreams(
const VideoCodec& codec,
const VideoOptions& options,
int max_bitrate_bps,
size_t num_streams);
static std::vector<webrtc::VideoStream> CreateSimulcastVideoStreams(
const VideoCodec& codec,
const VideoOptions& options,
int max_bitrate_bps,
size_t num_streams);
void* ConfigureVideoEncoderSettings(const VideoCodec& codec)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
void DestroyVideoEncoder(AllocatedEncoder* encoder)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
void SetCodec(const VideoCodecSettings& codec)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
void ReconfigureEncoder() EXCLUSIVE_LOCKS_REQUIRED(lock_);
bool 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() EXCLUSIVE_LOCKS_REQUIRED(lock_);
void UpdateHistograms() const EXCLUSIVE_LOCKS_REQUIRED(lock_);
rtc::ThreadChecker thread_checker_;
rtc::AsyncInvoker invoker_;
rtc::Thread* worker_thread_;
const std::vector<uint32_t> ssrcs_;
const std::vector<SsrcGroup> ssrc_groups_;
webrtc::Call* const call_;
rtc::VideoSinkWants sink_wants_;
// Counter used for deciding if the video resolution is currently
// restricted by CPU usage. It is reset if |source_| is changed.
int cpu_restricted_counter_;
// Total number of times resolution as been requested to be changed due to
// CPU adaptation.
int number_of_cpu_adapt_changes_;
// Total number of frames sent to |stream_|.
int frame_count_ GUARDED_BY(lock_);
// Total number of cpu restricted frames sent to |stream_|.
int cpu_restricted_frame_count_ GUARDED_BY(lock_);
rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
WebRtcVideoEncoderFactory* const external_encoder_factory_
GUARDED_BY(lock_);
rtc::CriticalSection lock_;
webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
// 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_ GUARDED_BY(lock_);
// 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_ GUARDED_BY(lock_);
bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
VideoFrameInfo last_frame_info_ GUARDED_BY(lock_);
bool sending_ GUARDED_BY(lock_);
// The timestamp of the first frame received
// Used to generate the timestamps of subsequent frames
rtc::Optional<int64_t> first_frame_timestamp_ms_ GUARDED_BY(lock_);
// The timestamp of the last frame received
// Used to generate timestamp for the black frame when source is removed
int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
};
// Wrapper for the receiver part, contains configs etc. that are needed to
// reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
// between rtc::VideoSinkInterface<webrtc::VideoFrame> and
// rtc::VideoSinkInterface<cricket::VideoFrame>.
class WebRtcVideoReceiveStream
: public rtc::VideoSinkInterface<webrtc::VideoFrame> {
public:
WebRtcVideoReceiveStream(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoReceiveStream::Config config,
WebRtcVideoDecoderFactory* external_decoder_factory,
bool default_stream,
const std::vector<VideoCodecSettings>& recv_codecs,
bool red_disabled_by_remote_side);
~WebRtcVideoReceiveStream();
const std::vector<uint32_t>& GetSsrcs() const;
rtc::Optional<uint32_t> GetFirstPrimarySsrc() 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<cricket::VideoFrame>* sink);
VideoReceiverInfo GetVideoReceiverInfo(bool log_stats);
// Used to disable RED/FEC when the remote description doesn't contain those
// codecs. This is needed to be able to work around an RTX bug which is only
// happening if the remote side doesn't send RED, but the local side is
// configured to receive RED.
// TODO(holmer): Remove this after a couple of Chrome versions, M53-54
// time frame.
void SetFecDisabledRemotely(bool disable);
private:
struct AllocatedDecoder {
AllocatedDecoder(webrtc::VideoDecoder* decoder,
webrtc::VideoCodecType type,
bool external);
webrtc::VideoDecoder* decoder;
// Decoder wrapped into a fallback decoder to permit software fallback.
webrtc::VideoDecoder* external_decoder;
webrtc::VideoCodecType type;
bool external;
};
void RecreateWebRtcStream();
void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs,
std::vector<AllocatedDecoder>* old_codecs);
AllocatedDecoder CreateOrReuseVideoDecoder(
std::vector<AllocatedDecoder>* old_decoder,
const VideoCodec& codec);
void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders);
std::string GetCodecNameFromPayloadType(int payload_type);
webrtc::Call* const call_;
StreamParams stream_params_;
webrtc::VideoReceiveStream* stream_;
const bool default_stream_;
webrtc::VideoReceiveStream::Config config_;
bool red_disabled_by_remote_side_;
WebRtcVideoDecoderFactory* const external_decoder_factory_;
std::vector<AllocatedDecoder> allocated_decoders_;
rtc::CriticalSection sink_lock_;
rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ 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_
GUARDED_BY(sink_lock_);
int64_t first_frame_timestamp_ 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_ GUARDED_BY(sink_lock_);
};
void Construct(webrtc::Call* call, WebRtcVideoEngine2* 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);
std::vector<VideoCodecSettings> FilterSupportedCodecs(
const std::vector<VideoCodecSettings>& mapped_codecs) const;
static bool ReceiveCodecsHaveChanged(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);
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_
GUARDED_BY(stream_crit_);
std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
GUARDED_BY(stream_crit_);
std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_);
std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_);
rtc::Optional<VideoCodecSettings> send_codec_;
rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;
WebRtcVideoEncoderFactory* const external_encoder_factory_;
WebRtcVideoDecoderFactory* const external_decoder_factory_;
std::vector<VideoCodecSettings> recv_codecs_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
webrtc::Call::Config::BitrateConfig 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_;
bool red_disabled_by_remote_side_;
int64_t last_stats_log_ms_;
};
} // namespace cricket
#endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_