Digital audio and/or video streaming system

A method for digital multimedia video and audio streaming including video and audio, for use with an encoder, a server, and a player, comprising the steps of:

• in an encoder:
• receiving video frames using a DirectX layer, via a video capture card, and simultaneously receiving audio signals in PCM samples via an audio input;
• for each encoder, converting the sampled audio and video signals into data objects respectively, along with the capture characteristics consisting of at least one of sample rate, bits and channels for audio and x, y and color space for video;
• encoding the converted data into encoded data, each encoder producing a view of the sample compatible with its input by converting and re-sampling the input data;
• partitioning the encoded data into smaller data packets;
• creating and attaching time-stamps to respective data packets, and, as a function of a transport mode, creating at least one of unicast RTP/UDP or TCP packets or multicast packets for transmission;
• in a server:
• determining, as a function of a network configuration of a network node on which the player is running, determine the type of network transport (RTP/UDP or TCP/Tunneled HTTP) and routing method (multicast or unicast) for the player;
• sending the data packets to all players that are connected thereto;
• in a player:
• placing each received packet in a sorted queue, a packet with one of an earliest time-stamp or a smallest sequence number being a first data packet in the queue;
• selecting the first packet out of the queue, copying the first packet to a synch buffer, and processing the first packet as follows:
• if a sleep time is less than 10 ms, processing the sample immediately;
• if the sleep time is greater than 50 ms, processing the sample after a 50 ms wait;
• if the sleep time is between 10 ms and 50 ms, sleeping for a predetermined number of milliseconds and then processing the sample;
• decoding each received frame, adding via a ring buffer a small audio lead time, and keeping one video frame in a buffer for a repaint;
• clearing, in response to a new audio frame, the ring buffer when the ring buffer is full, a new video frame replacing a previous video frame;
• feeding decoded frames to DirectX to be played back;
• updating the video frames, and playing back the audio stream;
• and
• sending an outgoing IRC message, when there is an IRC message to be sent, to an IRC server, and, when there are incoming IRC messages, displaying the IRC messages.

A digital audio streaming system, comprising:

• an encoder having an input port that receives an input digital audio signal and an output port that outputs an encoded digital audio signal, the encoded digital audio signal being encoded from the input digital signal;
• a server having at least one input port, which receives the encoded digital audio signal, operatively connected to the output port of the encoder from the encoder and at least one output port that outputs the encoded digital audio signal;
• and
• a player having an input port, which receives the encoded digital audio signal, operatively connected to the output port of the server, and an output port that outputs an output digital audio signal, the output digital signal being decoded from the encoded digital audio signal, the latency between the input digital audio signal and the output digital audio signal being less than one second.

A digital audio streaming system, comprising:

• an encoder having an input port that receives an input digital audio signal and an output port that outputs an encoded digital audio signal, the encoded digital audio signal being encoded from the input digital signal, the encoder having a first latency;
• a server having at least one input port, which receives the encoded digital audio signal, operatively connected to the output port of the encoder from the encoder and at least one output port that outputs the encoded digital audio signal, the server having a second latency;
• at least one player having an input port, which receives the encoded digital audio signal, operatively connected to the output port of the server, and an output port that outputs an output digital audio signal, the output digital signal being decoded from the encoded digital audio signal, the player having a third latency;
• and
• a system latency between a sum of the first, second, and third latencies, the system latency being less than one second.

A digital audio streaming system, comprising:

• an encoder having an input port that receives an input digital audio signal and an output port that outputs an encoded digital audio signal, the encoded digital audio signal being encoded from the input digital signal, the output port being operatively connected to the internet and the encoded digital signal being output to the internet in packets;
• a server having at least one input port, which receives the packets from the encoder, operatively connected to the output port of the encoder via the internet, and at least one output port that outputs the encoded digital audio signal;
• at least one player having an input port, which receives the encoded digital audio signal from the server, operatively connected to the output port of the server, and an output port that outputs an output digital audio signal, the output digital signal being decoded from the encoded digital audio signal;
• and
• a system latency between the input digital audio signal received by the encoder and the output digital audio signal output by the player being less than one second.

A method for digital audio streaming, comprising the steps of:

• in an encoder:
• waiting for the phone to ring;
• picking up, when a call is made, via a modem program of the encoder, the phone;
• recording 8 kHz PCM samples from speech input generated from the modem to produce audio signals;
• dividing the audio signals into 20 ms long frames;
• using a GSM codec to compress the 20 ms long frame into a data packet representing particular excitation sequence and amplitude by using short-term and long-term predictors;
• and
• time-stamping the packet with a current time;
• in a server:
• depending on the network configuration of the network node the player resides in, determining the type of network transport (RTP/UDP or TCP/Tunneled HTTP) and routing method (multicast or unicast) for the player;
• and
• sending the data packets to all the players that are connected to the server;
• in a player:
• placing each received packet in a sorted queue, a packet with one of an earliest time-stamp or a smallest sequence number being a first data packet in the queue;
• selecting the first packet out of the queue, coping the first packet to a synch buffer, and processing the first packet as follows:
• if a sleep time is less than 10 ms, processing the sample immediately;
• if the sleep time is greater than 50 ms, processing the sample after a 50 ms wait;
• if the sleep time is between 10 ms and 50 ms, sleeping for a predetermined number of milliseconds and then processing the sample;
• decoding each received frame, a ring buffer adding a small audio lead-time;
• clearing, in response to a new audio frame, the ring buffer when the ring buffer is full;
• feeding excitation signals in the frames through short-term and long-term synthesis filters to reconstruct the audio streams;
• and
• feeding the decoded audio streams to DirectX to be played back through a sound card.