call/video_send_stream.h - platform/external/webrtc - Gitiles

 /*
  *  Copyright (c) 2013 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 CALL_VIDEO_SEND_STREAM_H_
 #define CALL_VIDEO_SEND_STREAM_H_

 #include <map>
 #include <string>
 #include <utility>
 #include <vector>

 #include "api/call/transport.h"
 #include "api/rtpparameters.h"
 #include "api/rtp_headers.h"
 #include "api/videosinkinterface.h"
 #include "api/videosourceinterface.h"
 #include "call/rtp_config.h"
 #include "call/video_config.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "common_video/include/frame_callback.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/platform_file.h"

 namespace webrtc {

 class VideoEncoder;

 class VideoSendStream {
  public:
   struct StreamStats {
     StreamStats();
     ~StreamStats();

     std::string ToString() const;

     FrameCounts frame_counts;
     bool is_rtx = false;
     bool is_flexfec = false;
     int width = 0;
     int height = 0;
     // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
     int total_bitrate_bps = 0;
     int retransmit_bitrate_bps = 0;
     int avg_delay_ms = 0;
     int max_delay_ms = 0;
     StreamDataCounters rtp_stats;
     RtcpPacketTypeCounter rtcp_packet_type_counts;
     RtcpStatistics rtcp_stats;
   };

   struct Stats {
     Stats();
     ~Stats();
     std::string ToString(int64_t time_ms) const;
     std::string encoder_implementation_name = "unknown";
     int input_frame_rate = 0;
     int encode_frame_rate = 0;
     int avg_encode_time_ms = 0;
     int encode_usage_percent = 0;
     uint32_t frames_encoded = 0;
     uint32_t frames_dropped_by_capturer = 0;
     uint32_t frames_dropped_by_encoder_queue = 0;
     uint32_t frames_dropped_by_rate_limiter = 0;
     uint32_t frames_dropped_by_encoder = 0;
     rtc::Optional<uint64_t> qp_sum;
     // Bitrate the encoder is currently configured to use due to bandwidth
     // limitations.
     int target_media_bitrate_bps = 0;
     // Bitrate the encoder is actually producing.
     int media_bitrate_bps = 0;
     // Media bitrate this VideoSendStream is configured to prefer if there are
     // no bandwidth limitations.
     int preferred_media_bitrate_bps = 0;
     bool suspended = false;
     bool bw_limited_resolution = false;
     bool cpu_limited_resolution = false;
     bool bw_limited_framerate = false;
     bool cpu_limited_framerate = false;
     // Total number of times resolution as been requested to be changed due to
     // CPU/quality adaptation.
     int number_of_cpu_adapt_changes = 0;
     int number_of_quality_adapt_changes = 0;
     bool has_entered_low_resolution = false;
     std::map<uint32_t, StreamStats> substreams;
     webrtc::VideoContentType content_type =
         webrtc::VideoContentType::UNSPECIFIED;
     uint32_t huge_frames_sent = 0;
   };

   struct Config {
    public:
     Config() = delete;
     Config(Config&&);
     explicit Config(Transport* send_transport);

     Config& operator=(Config&&);
     Config& operator=(const Config&) = delete;

     ~Config();

     // Mostly used by tests.  Avoid creating copies if you can.
     Config Copy() const { return Config(*this); }

     std::string ToString() const;

     struct EncoderSettings {
       EncoderSettings() = default;
       EncoderSettings(std::string payload_name,
                       int payload_type,
                       VideoEncoder* encoder)
           : payload_name(std::move(payload_name)),
             payload_type(payload_type),
             encoder(encoder) {}
       std::string ToString() const;

       std::string payload_name;
       int payload_type = -1;

       // TODO(sophiechang): Delete this field when no one is using internal
       // sources anymore.
       bool internal_source = false;

       // Allow 100% encoder utilization. Used for HW encoders where CPU isn't
       // expected to be the limiting factor, but a chip could be running at
       // 30fps (for example) exactly.
       bool full_overuse_time = false;

       // Enables the new method to estimate the cpu load from encoding, used for
       // cpu adaptation.
       bool experiment_cpu_load_estimator = false;

       // Uninitialized VideoEncoder instance to be used for encoding. Will be
       // initialized from inside the VideoSendStream.
       VideoEncoder* encoder = nullptr;
     } encoder_settings;

     static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
     struct Rtp {
       Rtp();
       Rtp(const Rtp&);
       ~Rtp();
       std::string ToString() const;

       std::vector<uint32_t> ssrcs;

       // See RtcpMode for description.
       RtcpMode rtcp_mode = RtcpMode::kCompound;

       // Max RTP packet size delivered to send transport from VideoEngine.
       size_t max_packet_size = kDefaultMaxPacketSize;

       // RTP header extensions to use for this send stream.
       std::vector<RtpExtension> extensions;

       // See NackConfig for description.
       NackConfig nack;

       // See UlpfecConfig for description.
       UlpfecConfig ulpfec;

       struct Flexfec {
         Flexfec();
         Flexfec(const Flexfec&);
         ~Flexfec();
         // Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
         int payload_type = -1;

         // SSRC of FlexFEC stream.
         uint32_t ssrc = 0;

         // Vector containing a single element, corresponding to the SSRC of the
         // media stream being protected by this FlexFEC stream.
         // The vector MUST have size 1.
         //
         // TODO(brandtr): Update comment above when we support
         // multistream protection.
         std::vector<uint32_t> protected_media_ssrcs;
       } flexfec;

       // Settings for RTP retransmission payload format, see RFC 4588 for
       // details.
       struct Rtx {
         Rtx();
         Rtx(const Rtx&);
         ~Rtx();
         std::string ToString() const;
         // SSRCs to use for the RTX streams.
         std::vector<uint32_t> ssrcs;

         // Payload type to use for the RTX stream.
         int payload_type = -1;
       } rtx;

       // RTCP CNAME, see RFC 3550.
       std::string c_name;
     } rtp;

     struct Rtcp {
       Rtcp();
       Rtcp(const Rtcp&);
       ~Rtcp();
       std::string ToString() const;

       // Time interval between RTCP report for video
       int64_t video_report_interval_ms = 1000;
       // Time interval between RTCP report for audio
       int64_t audio_report_interval_ms = 5000;
     } rtcp;

     // Transport for outgoing packets.
     Transport* send_transport = nullptr;

     // Called for each I420 frame before encoding the frame. Can be used for
     // effects, snapshots etc. 'nullptr' disables the callback.
     rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;

     // Called for each encoded frame, e.g. used for file storage. 'nullptr'
     // disables the callback. Also measures timing and passes the time
     // spent on encoding. This timing will not fire if encoding takes longer
     // than the measuring window, since the sample data will have been dropped.
     EncodedFrameObserver* post_encode_callback = nullptr;

     // Expected delay needed by the renderer, i.e. the frame will be delivered
     // this many milliseconds, if possible, earlier than expected render time.
     // Only valid if |local_renderer| is set.
     int render_delay_ms = 0;

     // Target delay in milliseconds. A positive value indicates this stream is
     // used for streaming instead of a real-time call.
     int target_delay_ms = 0;

     // True if the stream should be suspended when the available bitrate fall
     // below the minimum configured bitrate. If this variable is false, the
     // stream may send at a rate higher than the estimated available bitrate.
     bool suspend_below_min_bitrate = false;

     // Enables periodic bandwidth probing in application-limited region.
     bool periodic_alr_bandwidth_probing = false;

     // Track ID as specified during track creation.
     std::string track_id;

    private:
     // Access to the copy constructor is private to force use of the Copy()
     // method for those exceptional cases where we do use it.
     Config(const Config&);
   };

   // Updates the sending state for all simulcast layers that the video send
   // stream owns. This can mean updating the activity one or for multiple
   // layers. The ordering of active layers is the order in which the
   // rtp modules are stored in the VideoSendStream.
   // Note: This starts stream activity if it is inactive and one of the layers
   // is active. This stops stream activity if it is active and all layers are
   // inactive.
   virtual void UpdateActiveSimulcastLayers(
       const std::vector<bool> active_layers) = 0;

   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;

   // Based on the spec in
   // https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
   // These options are enforced on a best-effort basis. For instance, all of
   // these options may suffer some frame drops in order to avoid queuing.
   // TODO(sprang): Look into possibility of more strictly enforcing the
   // maintain-framerate option.
   enum class DegradationPreference {
     // Don't take any actions based on over-utilization signals.
     kDegradationDisabled,
     // On over-use, request lower frame rate, possibly causing frame drops.
     kMaintainResolution,
     // On over-use, request lower resolution, possibly causing down-scaling.
     kMaintainFramerate,
     // Try to strike a "pleasing" balance between frame rate or resolution.
     kBalanced,
   };

   virtual void SetSource(
       rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
       const DegradationPreference& degradation_preference) = 0;

   // Set which streams to send. Must have at least as many SSRCs as configured
   // in the config. Encoder settings are passed on to the encoder instance along
   // with the VideoStream settings.
   virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;

   virtual Stats GetStats() = 0;

   // Takes ownership of each file, is responsible for closing them later.
   // Calling this method will close and finalize any current logs.
   // Some codecs produce multiple streams (VP8 only at present), each of these
   // streams will log to a separate file. kMaxSimulcastStreams in common_types.h
   // gives the max number of such streams. If there is no file for a stream, or
   // the file is rtc::kInvalidPlatformFileValue, frames from that stream will
   // not be logged.
   // If a frame to be written would make the log too large the write fails and
   // the log is closed and finalized. A |byte_limit| of 0 means no limit.
   virtual void EnableEncodedFrameRecording(
       const std::vector<rtc::PlatformFile>& files,
       size_t byte_limit) = 0;
   inline void DisableEncodedFrameRecording() {
     EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
   }

  protected:
   virtual ~VideoSendStream() {}
 };

 }  // namespace webrtc

 #endif  // CALL_VIDEO_SEND_STREAM_H_
	/*
	* Copyright (c) 2013 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 CALL_VIDEO_SEND_STREAM_H_
	#define CALL_VIDEO_SEND_STREAM_H_

	#include <map>
	#include <string>
	#include <utility>
	#include <vector>

	#include "api/call/transport.h"
	#include "api/rtpparameters.h"
	#include "api/rtp_headers.h"
	#include "api/videosinkinterface.h"
	#include "api/videosourceinterface.h"
	#include "call/rtp_config.h"
	#include "call/video_config.h"
	#include "common_types.h" // NOLINT(build/include)
	#include "common_video/include/frame_callback.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "rtc_base/platform_file.h"

	namespace webrtc {

	class VideoEncoder;

	class VideoSendStream {
	public:
	struct StreamStats {
	StreamStats();
	~StreamStats();

	std::string ToString() const;

	FrameCounts frame_counts;
	bool is_rtx = false;
	bool is_flexfec = false;
	int width = 0;
	int height = 0;
	// TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
	int total_bitrate_bps = 0;
	int retransmit_bitrate_bps = 0;
	int avg_delay_ms = 0;
	int max_delay_ms = 0;
	StreamDataCounters rtp_stats;
	RtcpPacketTypeCounter rtcp_packet_type_counts;
	RtcpStatistics rtcp_stats;
	};

	struct Stats {
	Stats();
	~Stats();
	std::string ToString(int64_t time_ms) const;
	std::string encoder_implementation_name = "unknown";
	int input_frame_rate = 0;
	int encode_frame_rate = 0;
	int avg_encode_time_ms = 0;
	int encode_usage_percent = 0;
	uint32_t frames_encoded = 0;
	uint32_t frames_dropped_by_capturer = 0;
	uint32_t frames_dropped_by_encoder_queue = 0;
	uint32_t frames_dropped_by_rate_limiter = 0;
	uint32_t frames_dropped_by_encoder = 0;
	rtc::Optional<uint64_t> qp_sum;
	// Bitrate the encoder is currently configured to use due to bandwidth
	// limitations.
	int target_media_bitrate_bps = 0;
	// Bitrate the encoder is actually producing.
	int media_bitrate_bps = 0;
	// Media bitrate this VideoSendStream is configured to prefer if there are
	// no bandwidth limitations.
	int preferred_media_bitrate_bps = 0;
	bool suspended = false;
	bool bw_limited_resolution = false;
	bool cpu_limited_resolution = false;
	bool bw_limited_framerate = false;
	bool cpu_limited_framerate = false;
	// Total number of times resolution as been requested to be changed due to
	// CPU/quality adaptation.
	int number_of_cpu_adapt_changes = 0;
	int number_of_quality_adapt_changes = 0;
	bool has_entered_low_resolution = false;
	std::map<uint32_t, StreamStats> substreams;
	webrtc::VideoContentType content_type =
	webrtc::VideoContentType::UNSPECIFIED;
	uint32_t huge_frames_sent = 0;
	};

	struct Config {
	public:
	Config() = delete;
	Config(Config&&);
	explicit Config(Transport* send_transport);

	Config& operator=(Config&&);
	Config& operator=(const Config&) = delete;

	~Config();

	// Mostly used by tests. Avoid creating copies if you can.
	Config Copy() const { return Config(*this); }

	std::string ToString() const;

	struct EncoderSettings {
	EncoderSettings() = default;
	EncoderSettings(std::string payload_name,
	int payload_type,
	VideoEncoder* encoder)
	: payload_name(std::move(payload_name)),
	payload_type(payload_type),
	encoder(encoder) {}
	std::string ToString() const;

	std::string payload_name;
	int payload_type = -1;

	// TODO(sophiechang): Delete this field when no one is using internal
	// sources anymore.
	bool internal_source = false;

	// Allow 100% encoder utilization. Used for HW encoders where CPU isn't
	// expected to be the limiting factor, but a chip could be running at
	// 30fps (for example) exactly.
	bool full_overuse_time = false;

	// Enables the new method to estimate the cpu load from encoding, used for
	// cpu adaptation.
	bool experiment_cpu_load_estimator = false;

	// Uninitialized VideoEncoder instance to be used for encoding. Will be
	// initialized from inside the VideoSendStream.
	VideoEncoder* encoder = nullptr;
	} encoder_settings;

	static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
	struct Rtp {
	Rtp();
	Rtp(const Rtp&);
	~Rtp();
	std::string ToString() const;

	std::vector<uint32_t> ssrcs;

	// See RtcpMode for description.
	RtcpMode rtcp_mode = RtcpMode::kCompound;

	// Max RTP packet size delivered to send transport from VideoEngine.
	size_t max_packet_size = kDefaultMaxPacketSize;

	// RTP header extensions to use for this send stream.
	std::vector<RtpExtension> extensions;

	// See NackConfig for description.
	NackConfig nack;

	// See UlpfecConfig for description.
	UlpfecConfig ulpfec;

	struct Flexfec {
	Flexfec();
	Flexfec(const Flexfec&);
	~Flexfec();
	// Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
	int payload_type = -1;

	// SSRC of FlexFEC stream.
	uint32_t ssrc = 0;

	// Vector containing a single element, corresponding to the SSRC of the
	// media stream being protected by this FlexFEC stream.
	// The vector MUST have size 1.
	//
	// TODO(brandtr): Update comment above when we support
	// multistream protection.
	std::vector<uint32_t> protected_media_ssrcs;
	} flexfec;

	// Settings for RTP retransmission payload format, see RFC 4588 for
	// details.
	struct Rtx {
	Rtx();
	Rtx(const Rtx&);
	~Rtx();
	std::string ToString() const;
	// SSRCs to use for the RTX streams.
	std::vector<uint32_t> ssrcs;

	// Payload type to use for the RTX stream.
	int payload_type = -1;
	} rtx;

	// RTCP CNAME, see RFC 3550.
	std::string c_name;
	} rtp;

	struct Rtcp {
	Rtcp();
	Rtcp(const Rtcp&);
	~Rtcp();
	std::string ToString() const;

	// Time interval between RTCP report for video
	int64_t video_report_interval_ms = 1000;
	// Time interval between RTCP report for audio
	int64_t audio_report_interval_ms = 5000;
	} rtcp;

	// Transport for outgoing packets.
	Transport* send_transport = nullptr;

	// Called for each I420 frame before encoding the frame. Can be used for
	// effects, snapshots etc. 'nullptr' disables the callback.
	rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;

	// Called for each encoded frame, e.g. used for file storage. 'nullptr'
	// disables the callback. Also measures timing and passes the time
	// spent on encoding. This timing will not fire if encoding takes longer
	// than the measuring window, since the sample data will have been dropped.
	EncodedFrameObserver* post_encode_callback = nullptr;

	// Expected delay needed by the renderer, i.e. the frame will be delivered
	// this many milliseconds, if possible, earlier than expected render time.
	// Only valid if \|local_renderer\| is set.
	int render_delay_ms = 0;

	// Target delay in milliseconds. A positive value indicates this stream is
	// used for streaming instead of a real-time call.
	int target_delay_ms = 0;

	// True if the stream should be suspended when the available bitrate fall
	// below the minimum configured bitrate. If this variable is false, the
	// stream may send at a rate higher than the estimated available bitrate.
	bool suspend_below_min_bitrate = false;

	// Enables periodic bandwidth probing in application-limited region.
	bool periodic_alr_bandwidth_probing = false;

	// Track ID as specified during track creation.
	std::string track_id;

	private:
	// Access to the copy constructor is private to force use of the Copy()
	// method for those exceptional cases where we do use it.
	Config(const Config&);
	};

	// Updates the sending state for all simulcast layers that the video send
	// stream owns. This can mean updating the activity one or for multiple
	// layers. The ordering of active layers is the order in which the
	// rtp modules are stored in the VideoSendStream.
	// Note: This starts stream activity if it is inactive and one of the layers
	// is active. This stops stream activity if it is active and all layers are
	// inactive.
	virtual void UpdateActiveSimulcastLayers(
	const std::vector<bool> active_layers) = 0;

	// Starts stream activity.
	// When a stream is active, it can receive, process and deliver packets.
	virtual void Start() = 0;
	// Stops stream activity.
	// When a stream is stopped, it can't receive, process or deliver packets.
	virtual void Stop() = 0;

	// Based on the spec in
	// https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
	// These options are enforced on a best-effort basis. For instance, all of
	// these options may suffer some frame drops in order to avoid queuing.
	// TODO(sprang): Look into possibility of more strictly enforcing the
	// maintain-framerate option.
	enum class DegradationPreference {
	// Don't take any actions based on over-utilization signals.
	kDegradationDisabled,
	// On over-use, request lower frame rate, possibly causing frame drops.
	kMaintainResolution,
	// On over-use, request lower resolution, possibly causing down-scaling.
	kMaintainFramerate,
	// Try to strike a "pleasing" balance between frame rate or resolution.
	kBalanced,
	};

	virtual void SetSource(
	rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
	const DegradationPreference& degradation_preference) = 0;

	// Set which streams to send. Must have at least as many SSRCs as configured
	// in the config. Encoder settings are passed on to the encoder instance along
	// with the VideoStream settings.
	virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;

	virtual Stats GetStats() = 0;

	// Takes ownership of each file, is responsible for closing them later.
	// Calling this method will close and finalize any current logs.
	// Some codecs produce multiple streams (VP8 only at present), each of these
	// streams will log to a separate file. kMaxSimulcastStreams in common_types.h
	// gives the max number of such streams. If there is no file for a stream, or
	// the file is rtc::kInvalidPlatformFileValue, frames from that stream will
	// not be logged.
	// If a frame to be written would make the log too large the write fails and
	// the log is closed and finalized. A \|byte_limit\| of 0 means no limit.
	virtual void EnableEncodedFrameRecording(
	const std::vector<rtc::PlatformFile>& files,
	size_t byte_limit) = 0;
	inline void DisableEncodedFrameRecording() {
	EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
	}

	protected:
	virtual ~VideoSendStream() {}
	};

	} // namespace webrtc

	#endif // CALL_VIDEO_SEND_STREAM_H_