MULTIMEDIA COMMUNICATION

INSTITUTO SUPERIOR TÉCNICO

Year 2012/2013 – 2nd Semester, Responsible: Prof. Fernando Pereira

1st Exam – 8th June 2013 (Saturday), 9am

 

The marks should be out before 10th June (Monday), 9am at the CMul Web page and the exam checking session will on the 11th June (Tuesday), 10am in room 0.73. 

The exam is 3 hours long. Answer all the questions in a detailed way, including all the computations performed and justifying well your answers.

Don’t get ‘trapped’ by any question; move forward to another question and return later. Good luck !

 

 

I (1.0 + 1.0 + 1.0 = 3 val.)

Consider a facsimile transmission using the READ coding method at 3200 bit/s for pages with 1000 lines, each line with 1728 samples. Consider also that, on average, 80% of the samples in each line are white.

a)      Assuming that

1.      the unidimensionally coded lines have an average compression factor of 10 for the back runs and 20 for the white runs

2.      the bidimensionally coded lines have an average compression factor of 20 for the back runs and 25 for the white runs

compute the global compression factor when a value of k equal to 3 is used to limit the propagation of channel errors. (R: 20.83)

b)     Assume now that there is the need to increase the MSLT (Minimum Scan Line Time) due to receiver limitations and this implies a reduction in 20% of the compression factors stated above (this means the values are now 80% of the values above). For this case, determine the maximum value of k that may be used if one needs to obtain decoding resynchronization, on average, at least once every 500 bits.  (R: 5)

c)      In general, identify two advantages and one drawback of using lossless source coding regarding lossy source coding.

 

II (0.5 + 0.5 + 0.5 + 1.0 + 1.0 val. = 3.5 val.)

Consider the JPEG standard to code photographic images with a 576×720 luminance resolution, 4:2:2 colour subsampling and 8 bit/sample.

a)      How many more luminance blocks than (total) chrominance blocks exist in this type of images.  (R: the same number)

b)     Determine the average number of bits per pixel (considering both the luminance and the chrominances) that are spent when coding this type of image with a global compression factor (for the luminance and the chrominances together) of 20. (R: 0.8 bit/pel)

c)      Determine the total number of bits that have to be spent to code an image if an average number of 4 DCT coefficients are coded per block and each coefficient costs, on average, 3 bits for the luminance and 2 bits for the chrominance; additionally consider that the EOB (End of Block) word costs 2 bits. (R:155520 bit)

d)     Why is it reasonable to say that the DCT representation involves a frequency interpretation of the image content?

e)      Explain a mechanism allowing to exploit some redundancy between neighboring blocks in JPEG coding ?

 

 

 

III (1.0 + 1.5 + 0.5 + 0.5 = 3.5 val.)

Consider a videotelephony communication using Recommendation ITU-T H.261. The video sequence is coded with a CIF spatial resolution and a frame rate of 12.5 Hz at a rate of 128 kbit/s.

The encoder processes sequentially the macroblocks in the GOBs, and the bits are uniformly generated in the time period that the encoder usually dedicates to encode each image. At the encoder, the bits wait for transmission at the encoder output buffer.

Knowing that the first image has used 15360 bits, the second image 20480 bit, and the third image 10240 bits, determine:

a)      The time instants at which the encoder has for sure generated all (coding) bits for the first, second and third images considering the first image is acquired at time instant 0 s. (R: 80, 160 and 240 ms)

b)     The minimum size of the encoder output buffer in order all bits above are transmitted without problems. (R: 15360 bit)

c)      The initial visualization delay associated to the system defined in b).  (R: 200 ms)

d)     The maximum number of bits that the 5th image may spent. (R: 20480 bit)

 

IV (1 + 1 + 0.5 + 1 = 3.5 val.)

Consider the MPEG-1 Audio standard to code audio content with 22 kHz bandwidth; assume reasonable compression factors and the most usual number of bits per sample.

a)      How many complete stereo music pieces, with a duration of 4 minutes, can we store in a 900 MBytes disk using the Layer 3 of the MPEG-1 Audio standard to code the music content with a transparent quality regarding CD music content. (R: 255)

b)     What is the maximum duration of each music piece that we can afford if we want to store 1000 musics in the same disk as above using a Layer 2 MPEG-1 Audio codec? (R: 40.91 s)

c)      Explain how would the maximum number of stored musics vary if we increase the audio bandwidth three times but the audio becomes mono and not anymore stereo. (R: number would become 2/3)

d)     Describe two main technical differences between the MPEG-1 Audio Layer 2 and Layer 3 codecs and the corresponding advantages.

 

  V (3 val.)

Consider using the MPEG-1 Video codec to code CIF (396 macroblocks) video information at 25 Hz to be stored in a CD. Assume that M=3 is used and the I get 3 times more bits than the P frames while the P frames get 4 times more bits than the B frames (always on average). The average number of bits per macroblock in a B frame is 50.

For the conditions above, determine the acceptable set of N values if it is requested that the video bitrate does not exceed 1.8 Mbit/s and the maximum access time does not exceed 400 ms. Assume that the reading rate is the same as the coding rate. (R: N=6 and M=3)

 

VI (1 + 1 + 0.5 + 0.5 + 0.5 = 3.5 val.)

Consider a DVB digital TV system.

a)      Knowing that a DVB solution may ‘insert’ 10 Mbit/s of total bitrate in a 8 MHz bandwidth channel, determine what would be the source bitrate that may be ‘inserted’ if all the system parameters stay the same with the exception of the channel coding ratio that goes from ½ to 1/3 and the modulation that goes from 8-PSK to 64-QAM. (R: 6.666 Mbit/s)

b)     Why is it essential in a Single Frequency Network that all transmitters send precisely the same data and do that well synchronized to transmit the same symbol at precisely the same time ? How do the transmitters obtain the necessary time reference ?

c)      What are the two main components of the channel coding solution in DVB-x2 ?

d)     What main parameter can be used to tune the correction capability of the channel coding solution and what does this parameter express ?

e)      What is the main reason justifying the availability of two channel coding block lengths in DVB-x2 ?