In the hands of portable electronics, two different audio worlds have created conflicts. Over the years, analog engineers have struggled to create solutions that perfectly handle voice, music playback and ring tones. This article examines current technology advancements and explores the trend of audio integration for smartphones.
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For a while, digital audio was clearly divided into two parts: Hi-Fi and call. Hi-Fi generally means stereo and 16-bit resolution with a sampling rate of 44.1 kHz, which is the original Compact Disc specification. On the other hand, the telephone call is mono and low resolution, and is usually digitized in 8-bit and 8kHZ. Different types of mixed signal ICs are suitable for different applications. Hi-Fi audio codec ICs quickly use multi-bit sigma-delta technology to improve sound quality, while the phone is still very simple, low data rate and low-cost converters limit the possibility of improving sound quality. These two codec chips have different interfaces. There are some data formats for Hi-Fi stereos on the market, the most popular of which is I2S (Inter-IC Sound). Telephone voice codec ICs typically use a pulse code modulation (PCM) interface. Strictly speaking, pulse code modulation has included most of the digital formats used today, including I2S; its original purpose was to distinguish between digital coding and analog techniques like frequency modulation. However, in digital telephony, pulse code modulation usually refers to a specific, mono data format that is not compatible with Hi-Fi stereo audio.
The rise of computer audio has created the rise of another interface. Although the sound quality requirements are quite similar to the long-lived consumer audio market, computers are required to play audio files (especially 8 kHz, 44.1 kHz, and 48 kHz) that are played at different sample rates. It is possible to convert the sampling rate in software, but the cost is too expensive. Therefore, the most widely adopted AC'97 standard gives this task to the codec chip, which makes it much more efficient to execute with dedicated hardware. AC'97 has actually become the industry standard for computer audio.
In the beginning, the portable system was still the original intention: portable CDs, mini-discs and MP3 players still use I2S digital-to-analog converters (DACs), mobile phones use pulse-coded modulation, while PDAs that support audio functions usually have The same AC'97 compiled code as the desktop computer. Therefore, it is no surprise that the first-generation composite system usually includes a telephone and PDA circuit distributed adjacent to the device, a pulse code modulated audio code chip controlled by a communication processor, and a Hi connected to the application processor. -Fi stereo (AC'97 or I2S) codec chip. However, the design of this codec chip does not take into account the interconnectivity of the two audio subsystems, or only minimal consideration. In this regard, discrete solid-state switches are typically inserted into the analog signal path, causing noise and harmonic distortion and occupying board space.
Figure 1: The first generation of smartphones has two separate audio subsystems (voice and Hi-Fi). In addition to these two codec chips, other discrete components are required for signal exchange and mixing.
Integration trend
Obviously, a solution for this application is highly anticipated. The idea of ​​"system-on-a-chip" (SoC) has led some companies to integrate stereo-to-analog converters or codec chips with other large ICs. However, this method does not produce the sound quality of the proprietary audio chip. The combination of power management and audio ICs seems to sacrifice sound quality because the rectifier typically injects noise into adjacent audio signal circuits. Integrating audio into digital ICs also faces the same problem, as true Hi-Fi components typically need to be optimized for mixed-signal applications in a 0.35-micron process, while digital logic has been scaled to 0.18-micron or less. For the two circuits coexisting on the same chip, if the performance of the non-analog circuit must be sacrificed, otherwise the entire IC will be built on a larger geometry, and the chip size will be expanded to an unacceptable level.
Speaker amplifiers are particularly difficult to integrate because they generate a lot of heat and require heat dissipation. Many integrated chips lack this capability and therefore cannot be considered a true "system-on-a-chip" (SoC) solution because an external speaker driver IC is required. Another common problem is that the analog input or output is not sufficient because the IC is always expected to be as small as possible. For a square package in which the pins are placed, such as a QFN (Quad Flat Package, Wireless) package, the length of each side is extended by approximately 1 mm to accommodate some additional pins, if the initial volume of the IC is large If this increase, it will lead to a surge in board space. For example, increasing from 5 x 5 mm to 6 x 6 mm increases the board area by 11 mm2, while if it is initially 10 x 10 mm, the board area increases to 21 mm2.
Figure 2: Block diagram of an integrated smart phone codec chip with an audio DAC, a stereo Hi-Fi DAC, and two ADCs that are routed to a pulse code modulation or I2S interface.
A dedicated audio IC can avoid these problems. By integrating other mixed-signal functions, such as touch digitization and integration of voice and Hi-Fi codec functions, the total number of chips can be reduced. Here, the audio code is integrated into the phone chipset, so it should be suitable for H-Fi codec chips with additional analog input, output and internal mixing. However, even in this case, two codec chips are needed to accommodate the wireless Bluetooth headset, since many Bluetooth codecs have a pulse code modulation interface.
There are many ways to integrate audio. Sharing analog-to-digital converters and digital-to-analog converters can reduce hardware costs, but can result in the inability to play or record two audio streams simultaneously. Providing a dedicated converter for each function overcomes this problem and extends battery life because phone-grade audio blocks can be designed with lower power than Hi-Fi. However, such a solution would increase the cost of silicon chips. The general compromise is to use a separate digital-to-analog converter, but share the analog-to-digital converter. So even if you can still play audio during a call (for example, you can hear the ringtone or music of the second call at the same time in Tonghua), but you can't record to the application processor during the call, this limitation is acceptable because In such a usage scenario, it is impossible for a user to see so much value. The power consumption of the analog to digital converter can be further reduced by stopping the power to one of the channels and operating the other channels at a lower sampling rate.
Figure 3: Only two input pins and one microphone bias pin can be used to connect both microphones to the codec chip.
The internal circuit blocks between the communication and application circuits can be shared, but the interface has no way. This is because each audio stream operates on an independent frequency circuit and has its own frequency frequency. As long as this is the case, the integrated smart phone codec chip requires both a pulse code modulation interface and a separate I2S or AC'97 link.
In fixed systems, the audio frequency is usually generated by a quartz oscillator. For example, AC'97 indicates that the codec chip that meets the specifications has an on-chip oscillator that is connected to an external 24.576 MHz (512 × 48 kHz) quartz, and I2S uses a sampling rate that is several times higher, usually sampling. The rate is 256. However, in the design of smart phones, considering the extra power consumption, board space and the cost of the frequency quartz body, designers often get Hi-Fi audio frequencies from other frequencies that have appeared on the board. Although this extra frequency portion needs to be implemented by a phase-locked loop (PLL), this solution is still more popular than an additional quartz oscillator because the low-power, low-noise phase-locked loop can be integrated into the mix at a very low price. Signal IC. The same applies to frequencies that may be required by other subsystems, such as the standard 27 MHz frequency of an MPEG decoder supporting video. For I2S, different sampling rates require different frequency frequencies. By simply multiplying word clock LPCLK (which is the sampling rate) by 256 or any other fixed number, the phase-locked loop can provide the correct frequency in any case. Component suppliers are therefore also inclined to integrate one or two phase-locked loops into the codec chip of their smartphones.
microphone
In smart phones, many of the most difficult design issues are related to microphones. It is usually necessary to consider at least two microphones: a built-in (internal) microphone and an external microphone as part of the headset. Other more internal microphones may be present to eliminate noise or stereo recording, while hands-free car kits may be connected to an external microphone. In addition to telephone calls, these microphones can also be used to record voice memos, or even video clips, under the control of the application processor.
In order to completely avoid off-chip switching, the smart phone codec chip must provide sufficient input, preferably with an independently adjustable gain and flexibility path to cover all usage scenarios. In addition to recording, the "Sidetone" function should also be provided. This adds a weak version of the microphone signal to the analog output, so people who use the headset to make a call can hear their voice. Insertion detection enables seamless switching between internal and external microphones when the headphones are plugged in or not connected.
Noise is another common consideration. The high frequency and digital parts of the circuit can cause interference, which is caused by the microphone signal coming from the PCB line, and this interference is amplified by the on-chip preamplifier. To avoid this problem, careful PCB layout is a key factor, and differentiated microphone input is another effective countermeasure. However, different inputs have their own layout requirements: the two sets of PCB traces must operate in parallel or adjacent, so any noise picked up by one set of lines will also appear in the other set of lines, so Eliminated in the microphone amplifier.
Noise reduction is another independent problem that requires two microphones to solve; one picks up the horn sound with background noise and the other picks up only background noise. Satisfactory results are usually not produced if the analog circuit is simply eliminated, because the phase and amplitude of the two noise signals will vary depending on the direction in which the noise is emitted. Digital signal processing is required here. However, the codec chip must perform this task by digitizing the two microphone signals.
Another type of noise occurs when used outdoors, that is, wind-cut noise, which is mostly limited to frequencies below 200 Hz, so that part of the noise can be eliminated with a high-pass filter. The simplest solution is to use a smaller coupling capacitor at the microphone input. However, this will make the microphone unable to be used for indoor music recording, resulting in no bass portion. This filtering is therefore not necessary for dual-purpose microphones. Incidentally, most audio analog-to-digital converters have built-in high-pass filters to eliminate DC bias from digital signals. The IC industry has customized this feature for mobile applications by making the corner frequency an option, some Hz for Hi-Fi, and others for 100Hz and 200Hz. Voice that supports wind cut filtering. Of course, analog and digital filtering can also be combined to create higher order filtering characteristics.
Headphones and Headsets (Headset)
A binaural headset that handles mobile phones also requires special analog circuitry. The most obvious first priority is to re-plan the path from the single-ear headphones or other speaker output signals when the binaural headphones are plugged in. Although sockets with integrated mechanical switches can do this, they are often cumbersome and expensive. Furthermore, the signal level used for the speaker may not be suitable for a binaural headset. Independent analog outputs for single-ear headphones, speakers, and binaural headphones with independent volume control solve this problem and allow for simpler sockets. Although a mechanical switch is still required, as long as the single-column, single-throw type, the switch with one end connected to the ground pin is sufficient, only one external pin is required for this jack. However, in a multimedia phone, the activation of this switch is not necessarily due to the insertion of the earphone; it may be just a headset that does not include a microphone. Therefore, the presence or absence of a microphone should be detected separately. For an electronic microphone, this can be achieved by sensing the bias current of the microphone, and if no current flows, there is no microphone insertion. Conversely, a large and unusual bias current is also evident: to avoid adding other contacts to a standard headset or a binaural headphone jack, a button used to receive an incoming call on a binaural headset (so-called The hook switch) usually shorts the microphone. As a result, the bias current increases, indicating that the hook switch has been pressed. By adding a current sensor to the on-chip microphone bias circuit, the smart phone codec can detect both situations and automatically take the correct action for individual situations.
speaker
The number of speakers and output power in mobile phones have been expanding recently. However, in the 1990s, single-sided headphones were the king. Modern clamshell phones are equipped with internal and external speakers, so that music can be played whether the phone is turned on or off. Support for stereo ringtones requires two external speakers, and the universal hands-free feature requires another "large" speaker for mobile phone standards in addition to small single-ear headphones. As with a microphone, providing a dedicated analog output for each speaker can bring many benefits of off-chip switching. Since the speaker amplifier consumes a large supply current, it reduces power consumption when it is not started. It becomes a very important thing. The mobile phone codec chip provides more and more detailed power management and can individually turn each output on or off to avoid any unnecessary battery power wastage. Furthermore, regulators in existing power management solutions typically do not provide the current required to drive the speaker volume fully open. Compiler-code chip vendors have responded to this problem by designing the on-chip speaker amplifier to operate directly from the battery's voltage (typically 4.2V for lithium-ion batteries) rather than the adjusted supply voltage. In general, although this approach does not save power, this speaker amplifier consumes power that would otherwise be consumed by the regulator, but removes the need to add a regulator.
ring
In the past few years, the complexity of ring tones has steadily increased, from only continuous to multi-tone melody, to the final WAV and MP3 music clips, almost any sound can be generated in stereo. Musical Instrument Digital Interface (MIDI) has become the standard for music ringtones, and some have developed low-power MIDI chips for this new application. Integrating this IC into the audio subsystem requires additional analog input on the codec chip. This additional input is also useful for connecting FM radio ICs and adding other features to the multimedia package. MIDI tone generation can of course be integrated into the codec chip. However, it is generally preferred to store the desired ringtone as an audio file and play it through the existing Hi-Fi digital-to-analog converter, so there is no correlation. For the IC industry of IP Silicon Intelligence, this appeal is not very attractive.
future development
So what will happen to the future of smart phone audio? Today's more eye-catching digital audio trends include stereo progression to multi-channel surround sound formats, and may be used recently for most PCs and notebooks. Azalia" Intel High Definition Audio standard. Not long ago, there were still many people who ridiculed the idea of ​​putting stereo speakers on the phone, but the current market situation proves that they are wrong. Similarly, Azalia's new features are not yet reasonable enough to cost it higher than AC'97. The battle between I2S and AC'97 is still going on. Some designers prefer the less complex I2S, others prefer the lower pin counts offered by AC'97 and can easily handle different sample rates. Just as many low-power CPUs currently used in smart phones offer dual-standard audio interfaces to meet the needs of different camps, these two standards may continue to coexist. Conversely, designing a codec chip that supports both standards is much more difficult because AC'97's VRA (adjustable rate audio) requires a different frequency technology than I2S, as well as a large number of additional digital circuits.
Successful integration of application and communications processors into a single digital device and the use of a single audio frequency will enable the combination of voice and Hi-Fi audio interfaces, and may later turn into a less complex codec chip of the past. . But for now, IC vendors are working to integrate other existing mixed-signal components into their audio codec chips, including touch functions, voltage regulators, and power management. The integrated imaging solution that has been prepared so far has not been able to integrate audio and video or video functions, but this is not an absolute fact. At the same time, audio functions continue to increase, such as 3D enhancement, graphic equalizer and dynamic compression, and the sound quality, power consumption and package size have also made great progress.
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