Abstract: A Voice-over-Internet-Protocol (VoIP) system has improved audio-buffer control. Voice captured by a microphone (mic) is loaded into mic buffers by the sound card and sent to a VoIP application. When a mic buffer arrives from the sound card, a speaker buffer manager is activated. Voice data extracted from incoming VoIP packets is loaded into a speaker buffer and sent to a speaker queue on the sound card for playback. A speaker-buffer count is kept and increased as each speaker buffer is sent to the sound card, and decreased as each empty speaker buffer is recycled from the sound card back to the VoIP application. As each mic buffer arrives, the speaker buffer manager compares the speaker-buffer count to upper and lower limits and sends zero, one, or two speaker buffers when the speaker-buffer count is above, between, or below the limits. Speaker-buffer latency and playback timing irregularities are reduced.

Abstract: A Voice-over-Internet-Protocol (VoIP) system has improved audio-buffer control. Voice captured by a microphone (mic) is loaded into mic buffers by the sound card and sent to a VoIP application. When a mic buffer arrives from the sound card, a speaker buffer manager is activated. Voice data extracted from incoming VoIP packets is loaded into a speaker buffer and sent to a speaker queue on the sound card for playback. A speaker-buffer count is kept and increased as each speaker buffer is sent to the sound card, and decreased as each empty speaker buffer is recycled from the sound card back to the VoIP application. As each mic buffer arrives, the speaker buffer manager compares the speaker-buffer count to upper and lower limits and sends zero, one, or two speaker buffers when the speaker-buffer count is above, between, or below the limits. Speaker-buffer latency and playback timing irregularities are reduced.

Abstract: A closed-loop voice-over-Internet-Protocol (VoIP) system has a local and a remote VOIP application. Each VOIP application monitors incoming packet arrival times and durations of audio data in the incoming packets to estimate bandwidth. The bandwidth estimates are forwarded to the other VOIP application. The forwarded bandwidth estimates are compared to a sending bandwidth. When the bandwidth estimate is above the sending bandwidth, compression and audio-frame decimation are reduced to improve voice quality. When the bandwidth estimate falls below the sending bandwidth, audio compression and decimation are increased to improve efficiency. Packet size can also be increased. Congestion estimates can also be sent with the audio data, causing packet transmission to pause until congestion ends. Incoming packet latencies are compared to a moving average to determine the congestion estimate, while bandwidth estimates are made by comparing packet audio duration to time between packet arrivals.

Abstract: A voice-over-Internet-Protocol (VoIP) application estimates bandwidth and congestion of the reception path to the VoIP application from a sending VoIP application. Packet arrivals are timed and the inter-packet delay is compared to the voice duration of the data contained in the more recent packet. When the inter-packet delay is longer than the voice duration the network is slowing and the bandwidth estimate is reduced. The bandwidth estimate is increased when inter-packet delay is smaller than the voice duration. Packet latencies are the difference in send and receive times and are compared to a moving average latency. When the current packet's latency is longer than the moving average, congestion is detected. When the current packet's latency equals the moving average, the network has recovered from congestion and the congestion estimate is reduced. Congestion and bandwidth estimates are added to packets sent out to provide feedback to the other VoIP application.

Abstract: In an improved system for receiving digital voice signals from a data network, a jitter buffer manager monitors packet arrival times, determines a time varying transit delay variation parameter and adaptively controls jitter buffer size in response to the variation parameter. A speed control module responds to a control signal from the jitter buffer manager by modifying the rate of data consumption from the jitter buffer, to compensate for changes in buffer size, preferably in a manner which maintains audio output with acceptable, natural human speech characteristics. Preferably, the manager also calculates average packet delay and controls the speed control module to adaptively align the jitter buffer's center with the average packet delay time.

Abstract: When an incoming call is received, a signal is sent to a computer that monitors the telephone line. If the telephone is not answered, the messaging system sends a signal and streaming audio signals to the computer. The signals are output over speakers connected to the computer.

Abstract: The invention includes a method and system for digitally and automatically adjusting the audio volume of digitized speech signals received over a network such as the internet. The method includes: estimating an average frame volume estimate (VE) for each frame of data; calculating from a plurality of successive frame volume estimates at least one moving average of the volume estimates; comparing at least one of the moving averages with a known desired level that is associated with a psychoacoustically desirable audio volume level; calculating, independently of any compression applied to the data frame during encoding, a digital gain factor based upon the results of the aforementioned comparison; and adjusting a volume level of the audio data based upon the digital gain factor. The system of the invention includes several modules, which could be executed by software run on a microprocessor, for carrying out the method of the invention.

Abstract: When an incoming call is received, a signal is sent to a computer that monitors the telephone line. If the telephone is not answered, the messaging system sends a signal and streaming audio signals to the computer. The signals are output over speakers connected to the computer.

Abstract: An automatic gain controller for a digitized audio signal, comprising a buffer with a plurality of subframes. Each subframe contains digitized data samples of the signal, the subframes including at least one future subframe and a current subframe. Signal processing means (such as a DSP) is coupled to the memory for controlling gain of the audio signal represented by the current subframe in the buffer. The signal processing means includes means to control gain of the data samples using a stored program for computing a plurality of mean signal level values from the plurality of subframes, each mean signal level value in the plurality corresponding to one of the subframes. The program includes means for causing decay of gain on the signal represented by the current subframe when a first set of the mean values are each below a low threshold signal level.