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/*
* Copyright (c) 2012 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.
*/
// This sub-API supports the following functionalities:
//
// - Creating and deleting VideoEngine instances.
// - Creating and deleting channels.
// - Connect a video channel with a corresponding voice channel for audio/video
// synchronization.
// - Start and stop sending and receiving.
#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
#include "webrtc/common_types.h"
namespace webrtc {
class Config;
class VoiceEngine;
// CpuOveruseObserver is called when a system overuse is detected and
// VideoEngine cannot keep up the encoding frequency.
class CpuOveruseObserver {
public:
// Called as soon as an overuse is detected.
virtual void OveruseDetected() = 0;
// Called periodically when the system is not overused any longer.
virtual void NormalUsage() = 0;
protected:
virtual ~CpuOveruseObserver() {}
};
struct CpuOveruseOptions {
CpuOveruseOptions()
: enable_capture_jitter_method(false),
low_capture_jitter_threshold_ms(20.0f),
high_capture_jitter_threshold_ms(30.0f),
enable_encode_usage_method(true),
low_encode_usage_threshold_percent(55),
high_encode_usage_threshold_percent(85),
low_encode_time_rsd_threshold(-1),
high_encode_time_rsd_threshold(-1),
enable_extended_processing_usage(true),
frame_timeout_interval_ms(1500),
min_frame_samples(120),
min_process_count(3),
high_threshold_consecutive_count(2) {}
// Method based on inter-arrival jitter of captured frames.
bool enable_capture_jitter_method;
float low_capture_jitter_threshold_ms; // Threshold for triggering underuse.
float high_capture_jitter_threshold_ms; // Threshold for triggering overuse.
// Method based on encode time of frames.
bool enable_encode_usage_method;
int low_encode_usage_threshold_percent; // Threshold for triggering underuse.
int high_encode_usage_threshold_percent; // Threshold for triggering overuse.
// TODO(asapersson): Remove options, not used.
int low_encode_time_rsd_threshold; // Additional threshold for triggering
// underuse (used in addition to
// threshold above if configured).
int high_encode_time_rsd_threshold; // Additional threshold for triggering
// overuse (used in addition to
// threshold above if configured).
bool enable_extended_processing_usage; // Include a larger time span (in
// addition to encode time) for
// measuring the processing time of a
// frame.
// General settings.
int frame_timeout_interval_ms; // The maximum allowed interval between two
// frames before resetting estimations.
int min_frame_samples; // The minimum number of frames required.
int min_process_count; // The number of initial process times required before
// triggering an overuse/underuse.
int high_threshold_consecutive_count; // The number of consecutive checks
// above the high threshold before
// triggering an overuse.
bool Equals(const CpuOveruseOptions& o) const {
return enable_capture_jitter_method == o.enable_capture_jitter_method &&
low_capture_jitter_threshold_ms == o.low_capture_jitter_threshold_ms &&
high_capture_jitter_threshold_ms ==
o.high_capture_jitter_threshold_ms &&
enable_encode_usage_method == o.enable_encode_usage_method &&
low_encode_usage_threshold_percent ==
o.low_encode_usage_threshold_percent &&
high_encode_usage_threshold_percent ==
o.high_encode_usage_threshold_percent &&
low_encode_time_rsd_threshold == o.low_encode_time_rsd_threshold &&
high_encode_time_rsd_threshold == o.high_encode_time_rsd_threshold &&
enable_extended_processing_usage ==
o.enable_extended_processing_usage &&
frame_timeout_interval_ms == o.frame_timeout_interval_ms &&
min_frame_samples == o.min_frame_samples &&
min_process_count == o.min_process_count &&
high_threshold_consecutive_count == o.high_threshold_consecutive_count;
}
};
struct CpuOveruseMetrics {
CpuOveruseMetrics()
: capture_jitter_ms(-1),
avg_encode_time_ms(-1),
encode_usage_percent(-1),
encode_rsd(-1),
capture_queue_delay_ms_per_s(-1) {}
int capture_jitter_ms; // The current estimated jitter in ms based on
// incoming captured frames.
int avg_encode_time_ms; // The average encode time in ms.
int encode_usage_percent; // The average encode time divided by the average
// time difference between incoming captured frames.
// TODO(asapersson): Remove metric, not used.
int encode_rsd; // The relative std dev of encode time of frames.
int capture_queue_delay_ms_per_s; // The current time delay between an
// incoming captured frame until the frame
// is being processed. The delay is
// expressed in ms delay per second.
};
class WEBRTC_DLLEXPORT VideoEngine {
public:
// Creates a VideoEngine object, which can then be used to acquire sub‐APIs.
static VideoEngine* Create();
static VideoEngine* Create(const Config& config);
// Deletes a VideoEngine instance.
static bool Delete(VideoEngine*& video_engine);
// Specifies the amount and type of trace information, which will be created
// by the VideoEngine.
static int SetTraceFilter(const unsigned int filter);
// Sets the name of the trace file and enables non‐encrypted trace messages.
static int SetTraceFile(const char* file_nameUTF8,
const bool add_file_counter = false);
// Installs the TraceCallback implementation to ensure that the VideoEngine
// user receives callbacks for generated trace messages.
static int SetTraceCallback(TraceCallback* callback);
protected:
VideoEngine() {}
virtual ~VideoEngine() {}
};
class WEBRTC_DLLEXPORT ViEBase {
public:
// Factory for the ViEBase sub‐API and increases an internal reference
// counter if successful. Returns NULL if the API is not supported or if
// construction fails.
static ViEBase* GetInterface(VideoEngine* video_engine);
// Releases the ViEBase sub-API and decreases an internal reference counter.
// Returns the new reference count. This value should be zero
// for all sub-API:s before the VideoEngine object can be safely deleted.
virtual int Release() = 0;
// Initiates all common parts of the VideoEngine.
virtual int Init() = 0;
// Connects a VideoEngine instance to a VoiceEngine instance for audio video
// synchronization.
virtual int SetVoiceEngine(VoiceEngine* voice_engine) = 0;
// Creates a new channel.
virtual int CreateChannel(int& video_channel) = 0;
// Creates a new channel grouped together with |original_channel|. The channel
// can both send and receive video. It is assumed the channel is sending
// and/or receiving video to the same end-point.
// Note: |CreateReceiveChannel| will give better performance and network
// properties for receive only channels.
virtual int CreateChannel(int& video_channel,
int original_channel) = 0;
// Creates a new channel grouped together with |original_channel|. The channel
// can only receive video and it is assumed the remote end-point is the same
// as for |original_channel|.
virtual int CreateReceiveChannel(int& video_channel,
int original_channel) = 0;
// Deletes an existing channel and releases the utilized resources.
virtual int DeleteChannel(const int video_channel) = 0;
// Registers an observer to be called when an overuse is detected, see
// 'CpuOveruseObserver' for details.
// NOTE: This is still very experimental functionality.
virtual int RegisterCpuOveruseObserver(int channel,
CpuOveruseObserver* observer) = 0;
// Sets options for cpu overuse detector.
virtual int SetCpuOveruseOptions(int channel,
const CpuOveruseOptions& options) = 0;
// Gets cpu overuse measures.
virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics) = 0;
// Registers a callback which is called when send-side delay statistics has
// been updated.
// TODO(holmer): Remove the default implementation when fakevideoengine.h has
// been updated.
virtual void RegisterSendSideDelayObserver(
int channel, SendSideDelayObserver* observer) {}
// Specifies the VoiceEngine and VideoEngine channel pair to use for
// audio/video synchronization.
virtual int ConnectAudioChannel(const int video_channel,
const int audio_channel) = 0;
// Disconnects a previously paired VideoEngine and VoiceEngine channel pair.
virtual int DisconnectAudioChannel(const int video_channel) = 0;
// Starts sending packets to an already specified IP address and port number
// for a specified channel.
virtual int StartSend(const int video_channel) = 0;
// Stops packets from being sent for a specified channel.
virtual int StopSend(const int video_channel) = 0;
// Prepares VideoEngine for receiving packets on the specified channel.
virtual int StartReceive(const int video_channel) = 0;
// Stops receiving incoming RTP and RTCP packets on the specified channel.
virtual int StopReceive(const int video_channel) = 0;
// Retrieves the version information for VideoEngine and its components.
virtual int GetVersion(char version[1024]) = 0;
// Returns the last VideoEngine error code.
virtual int LastError() = 0;
protected:
ViEBase() {}
virtual ~ViEBase() {}
};
} // namespace webrtc
#endif // #define WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_BASE_H_