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There is reason to believe that 2007 will be the last year to break through the barriers of videophone entry into the mass market. Making such predictions is based on the following factors: Broadband is widely used in the home, and now it has a penetration rate of more than 50% in some regions of Asia, Europe and North America; Moore's Law is constantly pushing forward the development of processor processing power, making processing Media processing algorithms that support complex operations that are needed to achieve reliable, high-quality full-motion video; battery technology and power management advances allow Wi-Fi-based devices to use standby time and talk time separately And hours to calculate, not minutes; the last point worth noting is that the industry's maturity and the increasingly sophisticated voice and video software solutions based on IP (referred to as V2IP) enable these.
Although videophones have been around for a long time, price and performance are still a stumbling block to the mass market solution. Even if we switch from analog videophone to digital IP, the limited price and lack of processing power will not be suitable for the mass market. With the popularity of wired and wireless networks, and high-performance voice/video processing technologies from Freescale, Renesas, TI and other chip vendors, these issues are quickly resolved. Therefore, we shift our focus to the fourth point mentioned above, namely software. When devices connected to IP are not working properly, we can place problems and concerns on this software solution. And in most cases it works fine.
Voice + video design based on IP software platform
Whether it's a phone or any personal communication and multimedia device, an attractive and reliable user experience must be provided to successfully generate a mainstream market. Therefore, the quality and reliability of wireless transmission is very important for Wi-Fi videophones. Fortunately, the IEEE 802.11 wireless LAN standard is constantly evolving and is constantly improving in terms of data rate, range and security, so we no longer need to care about this aspect.
It took only 18 to 24 months for the consumer electronics market to feel truly stable and reliable VoIP products. However, designing, developing, and producing voice-video (V2IP) phones that support Wi-Fi requires significant resources for software development, integration, and validation. We divide the entire solution into four key parts and then study each part:
1. Operating system and silicon platform;
2. Embedded voice + video based on IP architecture;
3. Application service layer;
4. Graphical User Interface (GUI).
Operating system and chip platform
We are seeing more and more manufacturers using embedded Linux as the basis for VoIP phone products. It has many advantages, including a familiar and rich software development environment for developers, but the most important thing is to help manufacturers reduce the total material cost. There are many providers (such as MontaVista) that offer a very stable, well-supported version of Linux for low-power consumer devices.
Looking at the architecture used in the first generation of videophones, we can see that different processors are used for voice, video, and system control functions. Because of the processing needs, processors optimized for dense media processing operations (digital signal processing or DSP) are generally employed. For example, a DSP is used to handle speech processing functions, including speech encoding/decoding, tone generation and detection, echo cancellation, and noise reduction; a DSP or dedicated coprocessor to handle video encoding and decoding; and an application processor to manage VoIP call control. Protocol and user interface (see Figure 1). This approach requires multiple programming models and development tool chains, which in turn leads to the need for larger development teams, increased training and additional costs.
Figure 1: The first generation of videophones requires three processors.
Since the introduction of the first generation of IP videophones, the processing power of general purpose application processors has increased to enable all of the voice processor tasks typically implemented in DSPs to be implemented by application processors. This is a very important advancement especially for the Wi-Fi videophone market, as the basic need for wirelessly connected devices is to reduce power consumption and maximize battery life.
Voice quality enhancement (line and acoustic echo cancellation) for VoIP codecs (G.711, G.729AB, G.723.1, iLBC), audio processing (DTMF, and call tone detection/generation) through code assembly and manual software optimization And jitter buffers, etc., and other similar features are now effectively implemented on the application processor. More and more application processors integrate hardware acceleration, and we can use hardware acceleration to handle video encoding and decoding (see Figure 2).
Figure 2: A new example of designing a videophone
The increased processing power of today's application processors allows us to use advanced operating system environments, such as embedded Linux, to effectively partition the control and media processing required in a V2IP system. This in turn makes software development using a single processor and toolchain much simpler, reducing costs by reducing one or more expensive DSPs.
The videophone will utilize one or more of the following compression algorithms: H.263, H.264 or MPEG-4. Among them, H.264 (also known as MPEG-4 AVC) is optimal in providing the lowest bit rate and high quality real-time video. The disadvantage is that H.264 needs higher processing power than H.263. After entering 2007, a cost-effective processor with sufficient capacity to handle H.264 will be common.
IP-based embedded voice + video
At the heart of V2IP design is embedded voice and video processing, as well as software units that control and manage system (architecture) data streams. OEMs and original design manufacturers (ODMs) have three options for developing V2IP architectures:
1. Establish a complete V2IP software architecture from zero;
2. Obtain device and software stack licenses for integration, validation, silicon migration and interoperability testing;
3. Get pre-integrated and validated third-party architecture from third parties.
Unless IP and networked software development are the core strengths of your organization, the fastest, lowest risk, and most cost-effective option is to obtain a third-party architecture license. Highly optimized solutions will emerge in a form that can be quickly integrated into end product designs. Finding technologies that provide all media processing algorithms and VoIP call control, combined with a flexible architecture, enables end product developers to focus on designing a high-performance value-added device. Because of the real-time nature of IP flows, a tightly integrated V2IP architecture is critical to ensuring reliable and stable voice and video communications. From a wide range of media processing libraries to a range of quality of service (QoS) and networking customers, the V2IP software architecture will ultimately determine the quality and performance of voice/video communications.
OEMs should take care to ensure that they implement a resilient VoIP architecture. Some architecture needs to have the appropriate VoIP codec selected and configured on the fly, and the media processing unit dynamically configured within the specified media channel. The architecture and its associated scheduler components must ensure that all algorithms required for the specified channel definition are executed within the allowed time period. Although in a single-channel system, the scheduling task of these algorithms is just a series of sequential calls to the appropriate algorithm, multi-channel systems provide more complex scenarios, in which case each channel may require a different VoIP code. The decoder, as well as some channels, require echo cancellation while other channels do not. Videophones are usually single-channel systems, although 3-party audio/video calls are generally supported.
Not to mention designing Wi-Fi videophones, designing a current VoIP phone also requires product differentiation and supporting the next generation of services and functions. Traditional VoIP phones provide basic "general quality" speech codecs such as G.711 and video compression using the H.263 standard. Before we discuss it further, it should be noted that these codecs are 100% capable of personal video conferencing calls and have been successfully implemented for many years.
However, in the current era of high fidelity and high resolution, to be popular with the public, the next generation of videophones must support broadband audio and advanced video compression technology. Both AMR-WB (G.722.2) audio technology and H.264 video compression technology have greatly improved the communication experience, providing a more lively communication between the two parties.
In addition to broadband, audio and higher resolution video, there are many technologies that can improve the reliability, performance and voice and video quality of IP communications for end users. Further, the following features require more competitive VoIP and V2IP solutions.
Audio Protocol/Voice Quality Enhancement:
1. G.711, G.723, G.726, G.729AB, G.723.1, iLBC;
2. Audio playback and recording;
3. Three-way calling with local audio mixing;
4. G.168 line echo cancellation;
5. Full-duplex acoustic echo cancellation (hands-free calling);
6. Tone generation/detection of the call process in a particular country;
7. A universal tone generator;
8. Gain control - automatic and manual mode;
9. DTMF detection/generation/relay;
10. Over/under sampling of 8, 16 and 24 kHz.
Video protocol: 1. H.263; 2. MPEG-4 simple class; 3. H.264; 4. Video playback and recording support.
NAT Traversal: 1. STUN customer; 2. TURN (STUN relay) client; 3. ICE.
High-fidelity VoIP and multimedia support: 1. G.722.2 (AMR-WB) codec; 2. Broadband AEC/AES; 3. MP3 decoding; 4. SP-MIDI decoding; 5. RTSP streaming client.
Seamless integration of applications and GUI
Once the system designer chooses a powerful architecture for voice and video processing, call setup, and NAT traversal, the key to the design shifts to designing and implementing the user experience to differentiate the product from other V2IP devices on the market.
Currently, the user experience is a reflection of many factors, from the quality of the key components used to build the device (such as the quality of microphones, speakers, cameras, and displays) to the ease of use of difficult-to-quantify user interfaces. Real-time personal communication devices use better display technology, and the GUI is becoming more and more important for the user experience. Currently, even the most basic VoIP phones offer a full color display GUI and offer lively menus, caller photo displays and instant messages.
The integration of the GUI with the embedded V2IP architecture is not straightforward. The biggest obstacle faced by most developers is that the types handled in the GUI and V2IP architecture are essentially different: the V2IP architecture focuses on fast-response, media-oriented real-time processing; the GUI reflects the rapid, user-driven event-driven processing.
A well-designed V2IP architecture will provide a powerful application programming interface (API) that requires very little GUI interaction. In particular, APIs typically only need to respond to events generated by users or the network. This segmentation avoids the uncomfortable combination of event-driven and real-time media processing units, enabling simple integration, allowing developers to focus on value-added intuitive GUI development.
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