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/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef MEDIA_CONDUIT_ABSTRACTION_
#define MEDIA_CONDUIT_ABSTRACTION_
#include "nsISupportsImpl.h"
#include "nsXPCOM.h"
#include "nsDOMNavigationTiming.h"
#include "mozilla/RefPtr.h"
#include "CodecConfig.h"
#include "VideoTypes.h"
#include "MediaConduitErrors.h"
#include "ImageContainer.h"
#include "webrtc/common_types.h"
namespace webrtc {
class I420VideoFrame;
}
#include <vector>
namespace mozilla {
/**
* Abstract Interface for transporting RTP packets - audio/vidoeo
* The consumers of this interface are responsible for passing in
* the RTPfied media packets
*/
class TransportInterface
{
protected:
virtual ~TransportInterface() {}
public:
/**
* RTP Transport Function to be implemented by concrete transport implementation
* @param data : RTP Packet (audio/video) to be transported
* @param len : Length of the media packet
* @result : NS_OK on success, NS_ERROR_FAILURE otherwise
*/
virtual nsresult SendRtpPacket(const void* data, int len) = 0;
/**
* RTCP Transport Function to be implemented by concrete transport implementation
* @param data : RTCP Packet to be transported
* @param len : Length of the RTCP packet
* @result : NS_OK on success, NS_ERROR_FAILURE otherwise
*/
virtual nsresult SendRtcpPacket(const void* data, int len) = 0;
NS_INLINE_DECL_THREADSAFE_REFCOUNTING(TransportInterface)
};
/**
* This class wraps image object for VideoRenderer::RenderVideoFrame()
* callback implementation to use for rendering.
*/
class ImageHandle
{
public:
explicit ImageHandle(layers::Image* image) : mImage(image) {}
const RefPtr<layers::Image>& GetImage() const { return mImage; }
private:
RefPtr<layers::Image> mImage;
};
/**
* 1. Abstract renderer for video data
* 2. This class acts as abstract interface between the video-engine and
* video-engine agnostic renderer implementation.
* 3. Concrete implementation of this interface is responsible for
* processing and/or rendering the obtained raw video frame to appropriate
* output , say, <video>
*/
class VideoRenderer
{
protected:
virtual ~VideoRenderer() {}
public:
/**
* Callback Function reportng any change in the video-frame dimensions
* @param width: current width of the video @ decoder
* @param height: current height of the video @ decoder
* @param number_of_streams: number of participating video streams
*/
virtual void FrameSizeChange(unsigned int width,
unsigned int height,
unsigned int number_of_streams) = 0;
/**
* Callback Function reporting decoded I420 frame for processing.
* @param buffer: pointer to decoded video frame
* @param buffer_size: size of the decoded frame
* @param time_stamp: Decoder timestamp, typically 90KHz as per RTP
* @render_time: Wall-clock time at the decoder for synchronization
* purposes in milliseconds
* @handle: opaque handle for image object of decoded video frame.
* NOTE: If decoded video frame is passed through buffer , it is the
* responsibility of the concrete implementations of this class to own copy
* of the frame if needed for time longer than scope of this callback.
* Such implementations should be quick in processing the frames and return
* immediately.
* On the other hand, if decoded video frame is passed through handle, the
* implementations should keep a reference to the (ref-counted) image object
* inside until it's no longer needed.
*/
virtual void RenderVideoFrame(const unsigned char* buffer,
size_t buffer_size,
uint32_t time_stamp,
int64_t render_time,
const ImageHandle& handle) = 0;
virtual void RenderVideoFrame(const unsigned char* buffer,
size_t buffer_size,
uint32_t y_stride,
uint32_t cbcr_stride,
uint32_t time_stamp,
int64_t render_time,
const ImageHandle& handle) = 0;
NS_INLINE_DECL_THREADSAFE_REFCOUNTING(VideoRenderer)
};
/**
* Generic Interface for representing Audio/Video Session
* MediaSession conduit is identified by 2 main components
* 1. Attached Transport Interface for inbound and outbound RTP transport
* 2. Attached Renderer Interface for rendering media data off the network
* This class hides specifics of Media-Engine implementation from the consumers
* of this interface.
* Also provides codec configuration API for the media sent and recevied
*/
class MediaSessionConduit
{
protected:
virtual ~MediaSessionConduit() {}
public:
enum Type { AUDIO, VIDEO } ;
virtual Type type() const = 0;
/**
* Function triggered on Incoming RTP packet from the remote
* endpoint by the transport implementation.
* @param data : RTP Packet (audio/video) to be processed
* @param len : Length of the media packet
* Obtained packets are passed to the Media-Engine for further
* processing , say, decoding
*/
virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len) = 0;
/**
* Function triggered on Incoming RTCP packet from the remote
* endpoint by the transport implementation.
* @param data : RTCP Packet (audio/video) to be processed
* @param len : Length of the media packet
* Obtained packets are passed to the Media-Engine for further
* processing , say, decoding
*/
virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len) = 0;
virtual MediaConduitErrorCode StopTransmitting() = 0;
virtual MediaConduitErrorCode StartTransmitting() = 0;
virtual MediaConduitErrorCode StopReceiving() = 0;
virtual MediaConduitErrorCode StartReceiving() = 0;
/**
* Function to attach transmitter transport end-point of the Media conduit.
* @param aTransport: Reference to the concrete teansport implementation
* When nullptr, unsets the transmitter transport endpoint.
* Note: Multiple invocations of this call , replaces existing transport with
* with the new one.
* Note: This transport is used for RTP, and RTCP if no receiver transport is
* set. In the future, we should ensure that RTCP sender reports use this
* regardless of whether the receiver transport is set.
*/
virtual MediaConduitErrorCode SetTransmitterTransport(RefPtr<TransportInterface> aTransport) = 0;
/**
* Function to attach receiver transport end-point of the Media conduit.
* @param aTransport: Reference to the concrete teansport implementation
* When nullptr, unsets the receiver transport endpoint.
* Note: Multiple invocations of this call , replaces existing transport with
* with the new one.
* Note: This transport is used for RTCP.
* Note: In the future, we should avoid using this for RTCP sender reports.
*/
virtual MediaConduitErrorCode SetReceiverTransport(RefPtr<TransportInterface> aTransport) = 0;
virtual bool SetLocalSSRC(unsigned int ssrc) = 0;
virtual bool GetLocalSSRC(unsigned int* ssrc) = 0;
virtual bool GetRemoteSSRC(unsigned int* ssrc) = 0;
virtual bool SetLocalCNAME(const char* cname) = 0;
/**
* Functions returning stats needed by w3c stats model.
*/
virtual bool GetVideoEncoderStats(double* framerateMean,
double* framerateStdDev,
double* bitrateMean,
double* bitrateStdDev,
uint32_t* droppedFrames) = 0;
virtual bool GetVideoDecoderStats(double* framerateMean,
double* framerateStdDev,
double* bitrateMean,
double* bitrateStdDev,
uint32_t* discardedPackets) = 0;
virtual bool GetAVStats(int32_t* jitterBufferDelayMs,
int32_t* playoutBufferDelayMs,
int32_t* avSyncOffsetMs) = 0;
virtual bool GetRTPStats(unsigned int* jitterMs,
unsigned int* cumulativeLost) = 0;
virtual bool GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
uint32_t* jitterMs,
uint32_t* packetsReceived,
uint64_t* bytesReceived,
uint32_t* cumulativeLost,
int32_t* rttMs) = 0;
virtual bool GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
unsigned int* packetsSent,
uint64_t* bytesSent) = 0;
virtual uint64_t CodecPluginID() = 0;
virtual void DeleteStreams() = 0;
NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaSessionConduit)
};
// Abstract base classes for external encoder/decoder.
class CodecPluginID
{
public:
virtual ~CodecPluginID() {}
virtual uint64_t PluginID() const = 0;
};
class VideoEncoder : public CodecPluginID
{
public:
virtual ~VideoEncoder() {}
};
class VideoDecoder : public CodecPluginID
{
public:
virtual ~VideoDecoder() {}
};
/**
* MediaSessionConduit for video
* Refer to the comments on MediaSessionConduit above for overall
* information
*/
class VideoSessionConduit : public MediaSessionConduit
{
public:
/**
* Factory function to create and initialize a Video Conduit Session
* return: Concrete VideoSessionConduitObject or nullptr in the case
* of failure
*/
static RefPtr<VideoSessionConduit> Create();
enum FrameRequestType
{
FrameRequestNone,
FrameRequestFir,
FrameRequestPli,
FrameRequestUnknown
};
VideoSessionConduit() : mFrameRequestMethod(FrameRequestNone),
mUsingNackBasic(false),
mUsingTmmbr(false),
mUsingFEC(false) {}
virtual ~VideoSessionConduit() {}
virtual Type type() const { return VIDEO; }
/**
* Function to attach Renderer end-point of the Media-Video conduit.
* @param aRenderer : Reference to the concrete Video renderer implementation
* Note: Multiple invocations of this API shall remove an existing renderer
* and attaches the new to the Conduit.
*/
virtual MediaConduitErrorCode AttachRenderer(RefPtr<VideoRenderer> aRenderer) = 0;
virtual void DetachRenderer() = 0;
/**
* Function to deliver a capture video frame for encoding and transport
* @param video_frame: pointer to captured video-frame.
* @param video_frame_length: size of the frame
* @param width, height: dimensions of the frame
* @param video_type: Type of the video frame - I420, RAW
* @param captured_time: timestamp when the frame was captured.
* if 0 timestamp is automatcally generated
* NOTE: ConfigureSendMediaCodec() MUST be called before this function can be invoked
* This ensures the inserted video-frames can be transmitted by the conduit
*/
virtual MediaConduitErrorCode SendVideoFrame(unsigned char* video_frame,
unsigned int video_frame_length,
unsigned short width,
unsigned short height,
VideoType video_type,
uint64_t capture_time) = 0;
virtual MediaConduitErrorCode SendVideoFrame(webrtc::I420VideoFrame& frame) = 0;
virtual MediaConduitErrorCode ConfigureCodecMode(webrtc::VideoCodecMode) = 0;
/**
* Function to configure send codec for the video session
* @param sendSessionConfig: CodecConfiguration
* @result: On Success, the video engine is configured with passed in codec for send
* On failure, video engine transmit functionality is disabled.
* NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
* transmission sub-system on the engine
*
*/
virtual MediaConduitErrorCode ConfigureSendMediaCodec(const VideoCodecConfig* sendSessionConfig) = 0;
/**
* Function to configurelist of receive codecs for the video session
* @param sendSessionConfig: CodecConfiguration
* NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
* reception sub-system on the engine
*
*/
virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
const std::vector<VideoCodecConfig* >& recvCodecConfigList) = 0;
/**
* Set an external encoder
* @param encoder
* @result: on success, we will use the specified encoder
*/
virtual MediaConduitErrorCode SetExternalSendCodec(VideoCodecConfig* config,
VideoEncoder* encoder) = 0;
/**
* Set an external decoder
* @param decoder
* @result: on success, we will use the specified decoder
*/
virtual MediaConduitErrorCode SetExternalRecvCodec(VideoCodecConfig* config,
VideoDecoder* decoder) = 0;
/**
* Function to enable the RTP Stream ID (RID) extension
* @param enabled: enable extension
* @param id: id to be used for this rtp header extension
* NOTE: See VideoConduit for more information
*/
virtual MediaConduitErrorCode EnableRTPStreamIdExtension(bool enabled, uint8_t id) = 0;
/**
* These methods allow unit tests to double-check that the
* max-fs and max-fr related settings are as expected.
*/
virtual unsigned short SendingWidth() = 0;
virtual unsigned short SendingHeight() = 0;
virtual unsigned int SendingMaxFs() = 0;
virtual unsigned int SendingMaxFr() = 0;
/**
* These methods allow unit tests to double-check that the
* rtcp-fb settings are as expected.
*/
FrameRequestType FrameRequestMethod() const {
return mFrameRequestMethod;
}
bool UsingNackBasic() const {
return mUsingNackBasic;
}
bool UsingTmmbr() const {
return mUsingTmmbr;
}
bool UsingFEC() const {
return mUsingFEC;
}
protected:
/* RTCP feedback settings, for unit testing purposes */
FrameRequestType mFrameRequestMethod;
bool mUsingNackBasic;
bool mUsingTmmbr;
bool mUsingFEC;
};
/**
* MediaSessionConduit for audio
* Refer to the comments on MediaSessionConduit above for overall
* information
*/
class AudioSessionConduit : public MediaSessionConduit
{
public:
/**
* Factory function to create and initialize an Audio Conduit Session
* return: Concrete AudioSessionConduitObject or nullptr in the case
* of failure
*/
static RefPtr<AudioSessionConduit> Create();
virtual ~AudioSessionConduit() {}
virtual Type type() const { return AUDIO; }
/**
* Function to deliver externally captured audio sample for encoding and transport
* @param audioData [in]: Pointer to array containing a frame of audio
* @param lengthSamples [in]: Length of audio frame in samples in multiple of 10 milliseconds
* Ex: Frame length is 160, 320, 440 for 16, 32, 44 kHz sampling rates
respectively.
audioData[] is lengthSamples in size
say, for 16kz sampling rate, audioData[] should contain 160
samples of 16-bits each for a 10m audio frame.
* @param samplingFreqHz [in]: Frequency/rate of the sampling in Hz ( 16000, 32000 ...)
* @param capture_delay [in]: Approx Delay from recording until it is delivered to VoiceEngine
in milliseconds.
* NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked
* This ensures the inserted audio-samples can be transmitted by the conduit
*
*/
virtual MediaConduitErrorCode SendAudioFrame(const int16_t audioData[],
int32_t lengthSamples,
int32_t samplingFreqHz,
int32_t capture_delay) = 0;
/**
* Function to grab a decoded audio-sample from the media engine for rendering
* / playoutof length 10 milliseconds.
*
* @param speechData [in]: Pointer to a array to which a 10ms frame of audio will be copied
* @param samplingFreqHz [in]: Frequency of the sampling for playback in Hertz (16000, 32000,..)
* @param capture_delay [in]: Estimated Time between reading of the samples to rendering/playback
* @param lengthSamples [out]: Will contain length of the audio frame in samples at return.
Ex: A value of 160 implies 160 samples each of 16-bits was copied
into speechData
* NOTE: This function should be invoked every 10 milliseconds for the best
* peformance
* NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can be invoked
* This ensures the decoded samples are ready for reading.
*
*/
virtual MediaConduitErrorCode GetAudioFrame(int16_t speechData[],
int32_t samplingFreqHz,
int32_t capture_delay,
int& lengthSamples) = 0;
/**
* Function to configure send codec for the audio session
* @param sendSessionConfig: CodecConfiguration
* NOTE: See VideoConduit for more information
*/
virtual MediaConduitErrorCode ConfigureSendMediaCodec(const AudioCodecConfig* sendCodecConfig) = 0;
/**
* Function to configure list of receive codecs for the audio session
* @param sendSessionConfig: CodecConfiguration
* NOTE: See VideoConduit for more information
*/
virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
const std::vector<AudioCodecConfig* >& recvCodecConfigList) = 0;
/**
* Function to enable the audio level extension
* @param enabled: enable extension
* @param id: id to be used for this rtp header extension
* NOTE: See AudioConduit for more information
*/
virtual MediaConduitErrorCode EnableAudioLevelExtension(bool enabled, uint8_t id) = 0;
virtual bool SetDtmfPayloadType(unsigned char type) = 0;
virtual bool InsertDTMFTone(int channel, int eventCode, bool outOfBand,
int lengthMs, int attenuationDb) = 0;
};
}
#endif
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