COMUNICAÇÃO DE ÁUDIO E VÍDEO

INSTITUTO SUPERIOR TÉCNICO

                          Year 2012/2013 – 1st Semester, MEEC, Responsible: Prof. Fernando Pereira

  2nd Exam – 30th January 2013, 11.30am (Wednesday)

 

The marks should be out before the 31st January (Thursday), 5pm, at the CAV Web page and the exam checking session will be on the 31st January (Thursday), 6pm, in room LT4. 

The exam is 3 hours long. Answer all the questions in a detailed way, including all the computations performed and justifying all answers. Don’t get ‘trapped’ by any question; move forward to another question and return later. Good luck!

 

I (1 + 1 + 0.5 = 2.5 val.)

Consider a facsimile transmission using the Modified READ coding method, at 4800 bit/s, with 2000 lines pages, each line with 1728 samples. Assume that

1.      the unidimensionally coded lines have an average compression factor of 15 for the black runs and 20 for the white runs

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

a)      Determine the minimum percentage of white samples that each line must have to guarantee that the compression factor is larger than 22, knowing that due to the transmission errors no more than 9 bidimensionally coded lines may be used in sequence. (R: > 58%)

b)     Assuming now that the percentage of white samples is 70% (on average), determine the average transmission time for a page when coding is performed to assure the maximum resilience to errors. Determine the same time assuming the transmission has no errors and thus the coding is performed to maximize the compression factor. (R: 39.6 and 30.96 s)

c)      Explain the motivation for designing the Huffman coding for the black and white run lengths using terminating and make-up codes/words with lengths up to 64 and multiples of 64, respectively.

 

II (1 + 0.5 + 1 + 0.5) = 3.0 val.)

Consider a ITU-T H.261 videophone communication working with a video signal at QCIF spatial resolution (176×144 samples for the luminance, 4:2:0 chrominance subsampling, with 8 bit/sample), 10 Hz. Assume that the average global (luminance and chrominances) compression factor (measured over all the macroblocks in the image), without any external constraints in terms of coding modes is 25 (with header bits included).

a)      Explain what happens (and why) if, when the two videophones try to establish the communication, they find that one of them has CIF spatial resolution capabilities and the other only has QCIF resolution capabilities. (R: QCIF which is mandatory)

b)     Assuming that for each frame, on average, only 50 macroblocks generate code bits (the remaining ones are so similar to the previous image that no update is needed), determine the average overall compression factor (including luminance and chrominances) measured over the macroblocks which effectively generate code bits. (R: 12.63)

c)      For the situation in b), assume that to guarantee a higher error protection, one out of each 25 macroblocks spending bits is necessarily coded in intra mode. Assuming that the intra coding mode has a compression factor at the macroblock level which is half of the compression factor determined in b), determine what would be the global compression factor (considering all the macroblocks in the image) corresponding to this situation. (R: 24.04)

d)     If a transmission rate of 40 kbit/s is used, what would be maximum number of bits that a frame may spend if a maximum acquisition-visualization delay of 200 ms would be required ? Assume that the encoder generates the bits for each frame uniformly in the time period between the acquisition moments of each two successive frames. (R: 8000)

 

III (1 + 0.5 + 0.5 + 1 + 0.5 = 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 mono music pieces, with a duration of 3 minutes, is possible to store in a 500 GBytes 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: 378787)

b)     What is the maximum duration of each music piece that it is possible to afford if 500 000 musics must the stored in the same disk as above using a MPEG-1 Audio Layer 2 codec ? (R: 90.9 s)

c)      Explain by what percentage would the maximum number of stored musics increase/reduce regarding the situation in a) if the audio bandwidth is reduced to half but the audio becomes stereo and not anymore mono. (R: 0%)

d)     Does an MPEG-1 Audio Layer 1 codec exploit more redundancy or irrelevancy in its coding process ? Why ? (R: irrelevancy)

e)      Explain what is the first reason in terms of processing flow contributing for MPEG-1 Audio Layer 2 having a larger coding delay than MPEG-1 Audio Layer 1. (R: audio frame 3 times longer)

 

  IV (1 + 1.5 + 0.5 + 1 = 4 val.)

Consider that your company is contacted to design a videoconference system between the various European premises of a pharmaceutical company. Your client pretends to use transmission lines with the minimum possible bitrate, providing the target quality, with a acquisition-visualization delay that should not exceed 250 ms. The spatial resolution is CIF (352×288 luminance samples), 4:2:0, at 10 Hz, with the usual number of bits per sample. Assume that you have available, offering the target quality, two solutions:

1.      H.261 based solution with average compression factors of 20 and 30 for the luminance and chrominance, respectively; the critical compression factors (for the images spending more bits) are 10 and 15 for the luminance and chrominance, respectively.

2.      MPEG-2 Video based solution with N = M = 2 with average compression factors of 20 and 30 for the luminance and chrominance, respectively, for the I frames, and 30 and 45 for the luminance and chrominance, respectively, for the P and B frames. The critical compression factors are 80% of the average compression factors.

Assume that the transmission rate is always the same as the coding rate.

a)      Determine the bitrate and acquisition-visualization delay for the H.261 based solution. (R: 540.67 kbit/s and 200 ms)

b)     Determine the bitrate and acquisition-visualization delay for the MPEG-2 based solution. (R: 450.56 kbit/s and 350 ms)

c)      Which solution would you recommend to your client considering the requirements defined above ? (R: H.261)

d)     If it is possible to move to N = M = 3 in the MPEG-2 Video solution, what are the main advantage and disadvantage of this change in the context of the design above ? (R: lower bitrate, increased delay)

 

V (0.5 + 0.5 + 1 + 1 = 3.0 val.)

Consider a DVB based digital TV system.

a)      Knowing that a DVB solution may ‘insert’ 10 Mbit/s of source rate in a 8 MHz bandwidth channel, determine what is the channel coding rate knowing that the transmitted rate is 3 times higher than the source rate. (R: 1/3

b)     Knowing that a DVB solution may ‘insert’ 10 Mbit/s of source rate in a 8 MHz bandwidth channel, what is the transmitted bitrate if all the system parameters stay the same with the exception of the channel coding ratio that goes from 1/3 to 1/4 and the modulation that goes from 8-PSK to 16-QAM. (R: 40 Mbit/s)

c)      What positive and negative happens in a DVB-T context if the modulation efficiency of each carrier is reduced while keeping all the remaining parameters, notably the total number of carriers ?

d)     What is the trade-off that led to the definition of the COFDM 2k and 8k modes in DVB-T ? What is the benefit of adding more modes like it has been done in DVB-T2 ?

 

VI (1 + 1 + 1 + 1 = 4.0 val.)

Consider the H.264/AVC standard and its SVC and MVC extensions.

a)      What is the main reason to include the Intra prediction tool in the H.264/AVC codec ? Why was the Constrained Intra Coding Mode defined ?

b)     What is the main reason to perform Variable Block-Size Motion Compensation in the H.264/AVC codec ? What is the trade-off involved when dealing with this tool ?

c)      What were the two main weaknesses in previous scalable video coding solutions that SVC addressed with relative success ?

d)     What does happen if a MVC compliant stream is provided to a H.264/AVC compliant decoder with the right profile ? Why ?