A method comprises receiving a first sequence of access units and a second sequence of access units; decoding at least one access unit of the first sequence of access units; decoding a first decodable access unit of the second sequence of access units; determining whether a next decodable access unit in the second sequence of access units can be decoded before an output time of the next decodable access unit in the second sequence of access units; and skipping decoding of the next decodable access unit based on determining that the next decodable access unit cannot be decoded before the at least one of the decoding time and the output time of the next decodable access unit.

Method and apparatus for video coding and decoding

Recently the 3rd Generation Partnership Project (3GPP) and the Moving Picture Experts Group (MPEG) specified Dynamic Adaptive Streaming over HTTP (DASH) to cope with the shortages in progressive HTTP based downloading and Real-time Transport Protocol (RTP) over the User Datagram Protocol (UDP), shortly RTP/UDP, based streaming. This paper investigates rate adaptation for the serial segment fetching method and the parallel segment fetching method in Content Distribution Network (CDN). The serial segment fetching method requests and receives segments sequentially whereas the parallel segment fetching method requests media segments in parallel. First, a novel rate adaptation metric is presented in this paper, which is the ratio of the expected segment fetch time (ESFT) and the measured segment fetch time to detect network congestion and spare network capacity quickly. ESFT represents the optimum segment fetch time determined by the media segment duration multiplied by the number of parallel HTTP threads to deliver media segments and the remaining duration to fetch the next segment to keep a certain amount of media time in the client buffer. Second, two novel rate adaptation algorithms are proposed for the serial and the parallel segment fetching methods, respectively, based on the proposed rate adaptation metric. The proposed rate adaptation algorithms use a step-wise switch-up and a multi-step switch-down strategy upon detecting the spare networks capacity and congestion with the proposed rate adaptation metric. To provide a good convergence in the representation level for DASH in CDN, a sliding window is used to measure the latest multiple rate adaptation metrics to determine switch-up. To decide switch-down, a rate adaptation metric is used. Each rate adaptation metric represents a reception of a segment/portion of a segment, which can be fetched from the different edge servers in CDN, hence it can be used to estimate the corresponding edge server bandwidth. To avoid buffer overflow due to a slight mismatch in the optimum representation level and bandwidth, an idling method is used to idle a given duration before sending the next segment. In order to solve the fairness between different clients who compete for bandwidth, the prioritized optimum segment fetch time is assigned to the newly joined clients. The proposed rate adaptation method does not require any transport layer information, which is not available at the application layer without cross layer communication. Simulation results show that the proposed rate adaptation algorithms for the serial and the parallel segment fetching methods quickly adapt the media bitrate to match the end-to-end network capacity, provide an advanced convergence and fairness between different clients and also effectively control buffer underflow and overflow for DASH in CDN. The reported simulation results demonstrate that the parallel rate adaptation outperforms the serial DASH rate adaptation algorithm with respect to achievable media bitrates while the serial rate adaptation is superior to the parallel DASH with respect to the convergence and buffer underflow frequency.

Rate adaptation for dynamic adaptive streaming over HTTP in content distribution network

A system for transmitting information to a display includes one or more content sources, a network, a content processor, and a display device. The content sources are each capable of transmitting content to a content router in first formats according to packet-based communication protocol. The network selectively couples content sources to the content processor. The content processor requests and receives content from a selected content source. The content processor then modifies the first formats and generates a video stream based on the received content and transmits the video stream to the display device. The display device then displays the video stream.

System and method for routing content

A method for processing media files includes receiving a request to download a first media file and, in response to receiving the request, generating a first signature that describes a first characteristic of the first media file. The method also includes determining whether the first signature matches any of a first plurality of stored signatures. The method also includes generating a second signature that describes a second characteristic of the first media file in response to determining that the first signature matches one of the first plurality of stored signatures. Additionally the method includes determining whether the second signature matches any of a second plurality of stored signatures and, in response to determining that the second signature matches one or more of the second plurality of stored signatures, denying the request to download the first media file.

System and method for monitoring content

A system for routing media content files having different format characteristics to a plurality of clients over a communication network includes a content translation router. The content translation router is capable of retrieving the content files. In addition, the content translation router includes processing modules for concurrently modifying content files having differing format characteristics to be compatible for transmission over the communication network for viewing at different ones of the clients.

System and method for processing content

In this paper we describe how progressive download and adaptive streaming can be combined into a simple and efficient streaming framework. Based on the MPEG-4 file format (MP4) we use HTTP for transport and argue that these two components are sufficient for specifying an open streaming architecture. The client selects appropriate chunks from the MP4 file to be transferred based on (1) the header information (i.e. the “moov” box) in the first part of the file and (2) his observation of network throughput. The framework is completely client driven which allows for better server scalability and reduces signalling overhead. We discuss architecture and implementation issues such as complexity, interoperability and scalability and compare to 3GPP PSS Rel-6 adaptive streaming when appropriate. Measurements from a simple MP4/HTTP streaming client are presented showing that appropriate chunks are selected such that increased reliability is achieved.

Adaptive progressive download based on the MPEG-4 file format

The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.

Low-frequency emphasis for lpc-based coding in frequency domain

An audio decoder decodes a bit stream of encoded audio data, wherein the bit stream of encoded audio data represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder comprises a determiner configured to determine whether a frame of the encoded audio data is a special frame comprising encoded audio sample values associated with the special frame and additional information, wherein the additional information comprise encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder. The decoder comprises an initializer configured to initialize the decoder, wherein initializing the decoder comprises decoding the encoded audio sample values included in the additional information before decoding the encoded audio sample values associated with the special frame.

Audio decoder, apparatus for generating encoded audio output data and methods permitting initializing a decoder

A time scaler for providing a time scaled version of an input audio signal is configured to compute or estimate a quality of a time scaled version of the input audio signal obtainable by a time scaling of the input audio signal. The time scaler is configured to perform the time scaling of the input audio signal in dependence on the computation or estimation of the quality of the time scaled version of the input audio signal obtainable by the time scaling. An audio decoder comprises such a time scaler.

Time scaler, audio decoder, method and a computer program using a quality control

The invention provides a decoder being configured for processing an encoded audio bitstream (BS), wherein the decoder (1 ) comprises: a bitstream decoder (2) configured to derive a decoded audio signal (DS) from the bitstream (BS), wherein the decoded audio signal (DS) comprises at least one decoded frame; a noise estimation device (3) configured to produce a noise estimation signal (NE) containing an estimation of the level and/or the spectral shape of a noise (N) in the decoded audio signal (DS); a comfort noise generating device (4) configured to derive a comfort noise signal (CN) from the noise estimation signal (NE); and a combiner (5) configured to combine the decoded frame of the decoded audio signal (DS) and the comfort noise signal (CN) in order to obtain an audio output signal (OS).

Comfort noise addition for modeling background noise at low bit-rates

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