MP3 Based Digital Audio Announcement For Mass Transit Systems

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1 MP3 Based Digital Audio Announcement For Mass Transit Systems Tobias Maisch INIT Innovations in Transportation, Inc. Chesapeake, VA Technology ABSTRACT One measure to make Mass Transit Systems more userfriendly and to comply with the requirements of the Americans With Disabilities Act (ADA) is to provide information to passengers. The name of the next stop is one of the most important types of information for passengers using public transit. This information is supplied with an audio and/or visible next stop information system. Digital audio announcements systems that are controlled by an onboard computer are a flexible and therefore commonly used solution. Digital audio announcement systems, however, often have the disadvantage of offering poor audio quality. The reason for the poor quality is that digital systems require a large storage capacity, yet storage is an expensive option. Therefore, smaller amounts of storage are often used with the result of unfavorable audio quality. With the use of audio compression algorithms based on digital signal processing such as MPEG II Layer 3 (MP3) a new technology is available. This new type of digital audio announcement system is able to combine excellent audio quality with only a moderate amount of storage and is therefore a very reasonable solution. The article will describe in brief the history and development of announcement technology in transit vehicles from announcements of operators / drivers to the newest MP3 based Digital Announcement System. Different digital announcement concepts and methods with pros and cons concerning audio quality, use of storage capacity, efforts for announcement production and integration in an Intelligent Transportation System (ITS) will be discussed. The articles will finally point out the benefits of the latest MP3 announcement systems and the integration in a Computer Aided Dispatch / Automatic Vehicle Location (CAD/AVL) System. REQUIREMEMTS Automatic audio announcement of the next stop information is a favorable way to make mass transit systems easier to use and to comply with the requirements of the ADA. The major requirements are: Excellent audio quality Moderate use of storage amount No additional workload for operator Easy way to produce and edit/modify recorded announcements Fast, easy and automated way to download new/ edited announcements from Control Center to vehicles with no or minimal workload for service personnel or operator HISTORY Next stop announcements of the vehicle operator was the original way to implement a next stop audio announcement system. But announcing the next stop or other information by vehicle operator had a lot of disadvantages. First, the workload of the driver was increased. He or she had to track the particular route and had to know at which point of his trip the announcements were to be made. Concentrating on driving could be inhibited. For this the danger of an accident was raised. Additionally, often the driver forgot to make the announcement or the announcement of the driver was not loud enough or clearly spoken. The satisfaction of vehicle operator and passenger of this system was low. The next step was to play back pre-recorded announcements from tape. In the first generation of this system the announcement had to be triggered from the vehicle operator and therefore he had to track his route in mind and had to know at which point of his trip he had to trigger the announcements. Further generations of tapebased announcement systems had an optional interface to AVL systems and the announcement was triggered by the AVL. The audio quality was good but the disadvantage of this system was the huge effort to maintain it. Tapes were wasted, the tape drive had to be cleaned regularly, and for a change of one announcement or the chronological order all tapes had to be re recorded again. Progress was made in the field of automated audio announcement systems by digitizing and storing the 1

2 announcement in digital semiconductor storage media for play back. Once the audio was digitized, editing, adding or changing the announcements and maintaining the whole system was much easier. The disadvantage of those systems was the huge amount of storage needed. To shorten the amount of storage to a moderate size different methods were used. All of these methods had the disadvantage of poor play back audio quality. With the use of digital signal processing algorithms like MP3 combined with the availability of huge processing performance (with embedded Digital Signal Processors), for the first time excellent audio quality in conjunction with moderate amount of storage is reached. CONCEPT AND ALGORITHMS OF DIGITAL ANNOUNCEMENT SYSTEMS Generally the whole system consists of three parts: Recording subsystem (see figure 1) Play back subsystem (see figure 2) Transfer Interface (Included in figure 1 and 2) RECORDING SUBSYSTEM Based on a PC with a high quality soundcard the recording subsystem digitize and quantize the audio signal with an Analog Digital Converter (ADC), integrated in the soundcard. Depending on the used sampling rate and resolution of the ADC the basics for an excellent audio quality will be set. This raw data, representing the recorded audio signal in digital format, is stored on hard disk for further editing like cutting, equalizing, level scaling, time scaling and so on. The processed and arranged audio announcements will be transformed to a suitable digital format through the next processing stage called coder. Concerning the amount of storage that will be needed for the final audio data and the achievable audio quality, the coder is the key element of the whole processing chain. After adding control and management data the announcement data is ready to transfer to the play back subsystem via the transfer interface. Throughout all steps of processing the announcements, the audio can be monitored by the audio output chain of the recording subsystem. It consists of the Digital Analog Converter (DAC) of the soundcard and the active Monitor Loudspeakers. PLAYBACK SUBSYSTEM FOR VEHICLES Based on a high sophisticated embedded hardware platform the playback subsystem consists of a storage unit, decoder, DAC and a control unit. This subsystem is connected to the Public Address system (PA) consisting of amplifier(s) and inside- and/or outside loudspeakers. Once the coded audio announcement data is loaded into the storage device, it is available to be played over the PA system. Therefore a command will be sent from the AVL system that knows the precise location of the vehicle to the control unit. After receiving a play back command, the decoder is fed with the coded data stream from the storage unit. The now decoded data stream is fed into the DAC where the digital data stream is converted into an analog signal, amplified and fed into the loudspeakers. AUDIO CODEC The word CODEC is an abbreviation made up of the two words CODER and DECODER. The task of a CODEC is to code and compress digitized raw audio data into a formatted data stream. The task of the DECODER is vice versa. It decodes and expands the formatted data stream back to the digitized raw audio data. There are many CODEC algorithms but the most popular are Pulse Code Modulation (PCM), (Adaptive) Difference Pulse Code Modulation (DPCM/ ADPCM) and MPEG II Layer 3 (MP3). A comparison of compression ratio, required bandwidth and amount of storage per second for the coded data stream is shown in table 1. PCM This CODEC only formats the raw digital audio data for transmission and storage. No compression will be carried out. Most popular example for PCM is the Compact Disk (CD). Music (audio data) stored on a CD is sampled with samples per second (44 ksps) and digitized with a resolution of 16 bit for each of the two stereo channels. Therefore the needed bandwidth is kbit/s (705.6 kbit/ s for each channel) and amount of storage of kbyte/s. For audio announcement systems the amount of storage is was high. Therefore the sampling rate and quantization was reduced. While reducing the quantization from 16 to 8 bit lead to a significant lower audio quality the reducing of the sampling rate was a compromise to lower the amount of needed storage. But the amount of needed memory was still to high for use in mass transit systems. DPCM /ADPCM While PCM is only formatting raw audio data, DPCM/ ADPCM is reducing the bandwidth of data stream by factor 2

3 2. This will be done by omitting all redundant data. Therefore the DPCM/ADPCM is using a prediction based on trend of the recent samples. Only the difference between the prediction and the real value of the sample is transmitted or stored. Because of the lower quantization that will be needed for coding the difference a reduction of bandwidth can be achieved. For better audio quality the quantization will be set dynamically when ADPCM is used. MP3 This CODEC is using a completely different algorithm (1,2,3,4). The raw audio data will be transformed from time domain to the frequency domain. The whole processing, transmission and storage of the data will be done in the frequency domain. To reach a compression factor of 10 to 14 an algorithm called perceptual noise shaping based on the psycho acoustic hearing model of humans is used. The perceptual part in the name means that the MP3 uses characteristics of the human ear to design the compression algorithm like: There are certain sounds that the human ear cannot hear There are certain sounds that the human ear hears much better than others If there are two sounds playing simultaneously we hear the louder one but cannot hear the softer one. This is called masking effect (figure 3). If there is a louder tone we can not her a softer one a short time (approx. 2-5 ms) before the beginning of the louder tone and a much longer time (up to 100ms) after ending of the louder tone. This is called pre masking and post masking effect (figure 4). Using facts like these about the human ear, certain parts of an announcement can be eliminated without significantly hurting the quality of the announcement for the listener. BENEFITS OF MP3 ANNOUNCEMENT SYSTEMS Based on an INIT Demonstration Installation of a MP3 announcement system (figure 5) that uses the INIT WIRELESSlan network for up and downloading MP3 files from the digital voice recording Workstation to the MP3 audio announcement system INIT MRI/MP3 there will be the following benefits: High Quality Audio Based on a 32 Mbyte Multi Media Card (MMC) for storing the MP3 files, there is place for ca sec. of near CD quality audio or 6500 sec. for FM radio quality audio. There are approximately 1300 and 2150 different, respectively, audio announcements in high audio quality. Worldwide standard MP3 is a worldwide standard that was first used for Internet applications but is now established in a broad range of audio recording, play back and storage application and devices. It is robust, proven und most important it is not proprietary. Off the Shelf Solution Using off-the-shelf components for digital voice recording workstation hardware, software, storage media like MMC and MP3 decoder chips a very competitive solution is available. Combing it with WIRELESSlan based on IEEE802.11b there is a transparent solution of automatic downloading of newest announcement files into the vehicle without any need of manual interaction. ACKNOWLEDGMENTS Thanks to Michael Wittemann who provided technical background. ENDNOTES 1. ISO/IEC :1993 Information technology Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s Part 3: Audio 2. ISO/IEC :1998 Information technology Generic coding of moving pictures and associated audio information Part 3: Audio (available in English only) 3. Home page of MPEG Group 4. : Homepage with MP3 basics 3

4 Figure 1. Structure of recording subsystem. 4

5 Figure 2. Structure of play back subsystem. Figure 3. Masking effect. 5

6 Figure 4. Pre and post maksing effect. Figure 5. Example of a full integrated audio announcement system. 6

7 Sampling rate (16 bit mono) [ksps] PCM Compression factor 1 Bandwidth [kbit/s] ([kbyte/s]) (88.2) (64.0) (44.1) Approx. Recordi ng time for 32 Mbyte [sec] DPCM/ADCPM Compression factor 2 Bandwidth [kbit/s] ([kbyte/s]) (44.1) (32.0) (22.05) Approx. Recordi ng time for 32 Mbyte [sec] MP3 Compression factor 10 to 14 Bandwidth [kbit/s] ([kbyte/s]) (8.0) (5.0) (4.0) Approx. Recording time for 32 Mbyte [sec] Table 1. Comparison of compression ratio and needed bandwidth of CODEC algorithms. 7

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