TL;DR: This paper uses a unique dataset that spans different content types, including short video on demand, long VoD, and live content from popular video con- tent providers, to measure quality metrics such as the join time, buffering ratio, average bitrate, rendering quality, and rate of buffering events.
Abstract: As the distribution of the video over the Internet becomes main- stream and its consumption moves from the computer to the TV screen, user expectation for high quality is constantly increasing. In this context, it is crucial for content providers to understand if and how video quality affects user engagement and how to best invest their resources to optimize video quality. This paper is a first step towards addressing these questions. We use a unique dataset that spans different content types, including short video on demand (VoD), long VoD, and live content from popular video con- tent providers. Using client-side instrumentation, we measure quality metrics such as the join time, buffering ratio, average bitrate, rendering quality, and rate of buffering events.We quantify user engagement both at a per-video (or view) level and a per-user (or viewer) level. In particular, we find that the percentage of time spent in buffering (buffering ratio) has the largest impact on the user engagement across all types of content. However, the magnitude of this impact depends on the content type, with live content being the most impacted. For example, a 1% increase in buffering ratio can reduce user engagement by more than three minutes for a 90-minute live video event. We also see that the average bitrate plays a significantly more important role in the case of live content than VoD content.
TL;DR: In this paper, a method and apparatus for allocating bandwidth within a bandwidth constrained interactive information distribution system was proposed, which determined if a requested information stream may be provided to a requesting subscriber at an appropriate bandwidth level (i.e., appropriate bitrate providing full quality), a minimal bandwidth level or not at all.
Abstract: A method and apparatus for allocating bandwidth within a bandwidth constrained interactive information distribution system. The system determined if a requested information stream may be provided to a requesting subscriber at an appropriate bandwidth level (i.e., appropriate bitrate providing full quality), a minimal bandwidth level (i.e., a reduced bitrate providing minimally acceptable quality) or not at all. Each information stream and any ancillary streams may be stored twice by the system, once at an appropriate encoded bitrate and once at a minimal encoded bitrate.
TL;DR: In this article, a hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes for each audio coding block, after a VBR quantization loop meets the NMR target, a second quantisation loop might be called to adaptively control the final bitrate that is not within a specified range.
Abstract: A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve
TL;DR: In this article, a method for optimizing data flow in a data network is presented, in which data packets from an application are to be transferred across the data network at a constant bitrate.
Abstract: The present invention relates to a method for optimizing data flow in a data network. Data that is to be transferred across the network at a constant bitrate is handled differently by a Transmission Control Protocol (TCP) layer compared to other data. A TCP congestion control process is adapted to recognize whether data packets from an application are to be transferred across the data network at a constant bitrate. The constant bitrate channel may be recognized by a Quality of Service identifier or a TCP port number. If the data packets do belong to the constant bitrate channel, the TCP congestion control limits the congestion window for these data packets. The congestion window for data packets other than those in the constant bitrate channel is allowed to increase, as usual, until data packets are lost, at which point the congestion window is reduced to a maximum segment size. The congestion window for constant bitrate packets, however, is increased until it reaches a maximum value. The limiting value of the congestion window is the constant bitrate multiplied by the round trip delay time between sending a packet and receiving its acknowledgement. The modifications to the TCP process may provide for the transmission of constant bitrate data over the data network that does not experience a timeout.