FUNDAMENTALS OF MULTIMEDIA ZE NIAN LI MARK S DREW PDF

About this book Introduction Multimedia is a ubiquitous part of the technological environment in which we work and think, touching upon almost all aspects of computer science and engineering. This comprehensive textbook introduces the Fundamentals of Multimedia in an accessible manner, addressing real issues commonly faced in the workplace. Suitable for both advanced undergraduate and graduate students, the essential concepts are explained in a practical way to enable students to apply their existing skills to address problems in multimedia. Fully revised and updated, this new edition now includes coverage of such topics as 3D TV, social networks, high-efficiency video compression and conferencing, wireless and mobile networks, and their attendant technologies. Topics and features: Presents a brief history and overview of the key concepts in multimedia, including important data representations and color science Reviews lossless and lossy compression methods for image, video and audio data Examines the demands placed by multimedia communications on wired and wireless networks Discusses the impact of social media and cloud computing on information sharing, and on multimedia content search and retrieval Includes study exercises at the end of each chapter Provides supplementary resources for both students and instructors at an associated website This classroom-tested textbook is ideal for higher-level undergraduate and graduate courses on multimedia systems.

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My old Soundblaster card is an 8—bit card. Answer: Quantization levels not sampling frequency. If a set of ear protectors reduces the noise level by 30 dB, how much do they reduce the intensity the power? Answer: A reduction in intensity of A loss of audio output at both ends of the audible frequency range is inevitable, due to the frequency response function of an audio amplifier and the medium e.

If the loss remains? Suppose the sampling frequency is 1. What is the alias frequency? Answer: 0. This is known as the cocktail-party effect. Theway it operates is that our hearing can localize a sound source by taking advantage of the difference in phase between the two signals entering our left and right ears binaural auditory perception. State how you think a karaoke machine works. That is, for an instrument, either the left or right channel is emphasized.

Answer: For the singer, left and right is always mixed with the exact same pan. This information can be used to subtract out the sound of the singer. To do so, replace the left channel by the difference between the left and the right, and boost the maximum amplitude; and similarly for the right channel. The dynamic range of a signal V is the ratio of the maximum to the minimum absolute value, expressed in decibels. The dynamic range expected in a signal is to some extent an expression of the signal quality.

It also dictates the number of bits per sample needed to reduce the quantization noise to an acceptable level. For example, we may want to reduce the noise to at least an order of magnitude below Vmin.

Suppose the dynamic range for a signal is 60 dB. Can we use 10 bits for this signal? Can we use 16 bits? Answer: The range is mapped to? In fact, whole range Vmax down to Vmax? The largest negative signal,? Vmax is mapped to? So this is not sufficient intensity resolution. Using uniform quantization, how many bits should we use to encode speech to make the quantization noise at least an order of magnitude less than the smallest detectable telephonic sound? Therefore to get quantization noise about a factor of 16 below the minimum sound, we need 12 bits.

Perceptual nonuniformity is a general term for describing the nonlinearity of human perception. That is, when a certain parameter of an audio signal varies, humans do not necessarily perceive the difference in proportion to the amount of change. Why could it improve quantization? It makes better use of the limited number of bits available for each quantized data.

Draw a diagram showing a sinusoid at 5. Draw the alias at 2. Suppose a signal contains tones at 1, 10, and 21 kHz and is sampled at the rate 12 kHz and then processed with an antialiasing filter limiting output to 6 kHz. What tones are included in the output? Hint: Most of the output consists of aliasing. Answer: No. If so, how is it done in MIDI? What does this message accomplish? Answer: Yes. Replaces patch for a channel. In terms of data, what is the main difference between the two types of messages?

Within those two categories, list the different subtypes. Answer: Channel Messages and System Messages. Channel voice messages, Channelmodemessages, System real-time messages, System common messages, System exclusive messages. Channel messages have a status byte with leading most-significant-bit set, and 4 bits of channel information; System messages have the 4 MSBs set. Answer: Changes the patch to 4 on channel 2.

In PCM, what is the delay, assuming 8 kHz sampling? Generally, delay is the time penalty associated with any algorithm due to sampling, processing, and analysis. Answer: Since there is no processing associated with PCM, the delay is simply the time interval between two samples, and at 8 kHz, this is 0.

If the input signal has values as follows: 20 38 56 74 92 show that the output from a DPCM coder without entropy coding is as follows: 20 44 56 74 89 Figure 6. As a programming project, write a small piece of code to verify your results. Show that on the decoder side we end up with reconstructed signal values as follows: 20 44 56 74 89 so that the error gets progressively worse.

Figure 6. Modify your code from above to verify this statement.

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