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64-1066: AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA , Wheatnet , Livewire , Q-LAN and Dante . AES67 promises interoperability between previously competing networked audio systems and long-term network interoperation between systems. It also provides interoperability with layer 2 technologies, like Audio Video Bridging (AVB) . Since its publication, AES67 has been implemented independently by several manufacturers and adopted by many others. AES67 defines requirements for synchronizing clocks, setting QoS priorities for media traffic, and initiating media streams with standard protocols from

128-529: A Technology & Engineering Emmy Award in 2019 for the development of synchronized multi-channel uncompressed audio transport over IP networks. The standard has been implemented by Lawo , Digisynthetic, Axia , AMX (in SVSI devices), Wheatstone, Extron Electronics , Riedel , Ross Video , ALC NetworX , Audinate , Archwave, Digigram, Sonifex, Aqua Broadcast, Yamaha , QSC , Neutrik, Attero Tech, Merging Technologies , Gallery SIENNA, Behringer , Tieline and

192-417: A coordination problem : it emerges from situations in which all parties realize mutual gains, but only by making mutually consistent decisions. Examples : Private standards are developed by private entities such as companies, non-governmental organizations or private sector multi-stakeholder initiatives, also referred to as multistakeholder governance . Not all technical standards are created equal. In

256-416: A voltage or current (depending on type) that represents the value presented on their digital inputs. This output would then generally be filtered and amplified for use. To recover the original signal from the sampled data, a demodulator can apply the procedure of modulation in reverse. After each sampling period, the demodulator reads the next value and transitions the output signal to the new value. As

320-410: A DS0 is either μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12- or 13-bit linear PCM sample number is mapped into an 8-bit value. This system is described by international standard G.711 . Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress

384-485: A NRZ system to be synchronized using in-band information, there must not be long sequences of identical symbols, such as ones or zeroes. For binary PCM systems, the density of 1-symbols is called ones-density . Ones-density is often controlled using precoding techniques such as run-length limited encoding, where the PCM code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into

448-422: A PTP clock can be designated as an ordinary clock (OC), boundary clock (BC) or transparent clock (TC), though 802.1AS transparent clocks also have some boundary clock capabilities. A device may implement one or more of these capabilities. OC may have as few as one port (network connection), while TC and BC must have two or more ports. BC and OC ports can work as a master (grandmaster) or a slave. An IEEE 1588 profile

512-459: A global sample frequency setting. Media packets are scheduled according to 'packet time' - transmission duration of a standard Ethernet packet. Packet time is negotiated by the stream source for each streaming session. Short packet times provide low latency and high transmission rate, but introduce high overhead and require high-performance equipment and links. Long packet times increase latencies and require more buffering. A range from 125 μs to 4 ms

576-434: A government (i.e., through legislation ), business contract, etc. The standardization process may be by edict or may involve the formal consensus of technical experts. The primary types of technical standards are: Technical standards are defined as: Technical standards may exist as: When a geographically defined community must solve a community-wide coordination problem , it can adopt an existing standard or produce

640-515: A large user base, doing some well established thing that between them is mutually incompatible. Establishing national/regional/international standards is one way of preventing or overcoming this problem. To further support this, the WTO Technical Barriers to Trade (TBT) Committee published the "Six Principles" guiding members in the development of international standards. The existence of a published standard does not imply that it

704-455: A larger aggregate data stream , generally for transmission of multiple streams over a single physical link. One technique is called time-division multiplexing (TDM) and is widely used, notably in the modern public telephone system. The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal. These devices are digital-to-analog converters (DACs). They produce

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768-412: A new one. The main geographic levels are: National/Regional/International standards is one way of overcoming technical barriers in inter-local or inter-regional commerce caused by differences among technical regulations and standards developed independently and separately by each local, local standards organisation , or local company. Technical barriers arise when different groups come together, each with

832-621: A number of papers in relation to the proliferation of private food safety standards in the agri-food industry, mostly driven by standard harmonization under the multistakeholder governance of the Global Food Safety Initiative (GFSI). With concerns around private standards and technical barriers to trade (TBT), and unable to adhere to the TBT Committee's Six Principles for the development of international standards because private standards are non-consensus,

896-589: A rate above 3500–4300 Hz; lower rates proved unsatisfactory. In 1920, the Bartlane cable picture transmission system used telegraph signaling of characters punched in paper tape to send samples of images quantized to 5 levels. In 1926, Paul M. Rainey of Western Electric patented a facsimile machine that transmitted its signal using 5-bit PCM, encoded by an opto-mechanical analog-to-digital converter . The machine did not go into production. British engineer Alec Reeves , unaware of previous work, conceived

960-552: A result of these transitions, the signal retains a significant amount of high-frequency energy due to imaging effects. To remove these undesirable frequencies, the demodulator passes the signal through a reconstruction filter that suppresses energy outside the expected frequency range (greater than the Nyquist frequency f s / 2 {\displaystyle f_{s}/2} ). Common sample depths for LPCM are 8, 16, 20 or 24 bits per sample . LPCM encodes

1024-562: A single international standard ; ISO 9001 (quality), ISO 14001 (environment), ISO 45001 (occupational health and safety), ISO 27001 (information security) and ISO 22301 (business continuity). Another example of a sector working with a single international standard is ISO 13485 (medical devices), which is adopted by the International Medical Device Regulators Forum (IMDRF). In 2020, Fairtrade International , and in 2021, Programme for

1088-505: A single sound channel. Support for multichannel audio depends on file format and relies on synchronization of multiple LPCM streams. While two channels (stereo) is the most common format, systems can support up to 8 audio channels (7.1 surround) or more. Common sampling frequencies are 48 kHz as used with DVD format videos, or 44.1 kHz as used in CDs. Sampling frequencies of 96 kHz or 192 kHz can be used on some equipment, but

1152-646: A time. Rather than natural binary, the grid of Goodall's later tube was perforated to produce a glitch-free Gray code and produced all bits simultaneously by using a fan beam instead of a scanning beam. In the United States, the National Inventors Hall of Fame has honored Bernard M. Oliver and Claude Shannon as the inventors of PCM, as described in "Communication System Employing Pulse Code Modulation", U.S. patent 2,801,281 filed in 1946 and 1952, granted in 1956. Another patent by

1216-456: Is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm ). Though PCM is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to

1280-613: Is achieved with IEEE 802.1AS profile for Time-Sensitive Applications. The media clock is based on synchronized network time with an IEEE 1588 epoch (1 January 1970 00:00:00 TAI). Clock rates are fixed at audio sampling frequencies of 44.1 kHz, 48 kHz and 96 kHz (i.e. thousand samples per second). RTP transport works with a fixed time offset to network clock. Media data is transported in IPv4 packets and attempts to avoid IP fragmentation . Real-time Transport Protocol with RTP Profile for Audio and Video (L24 and L16 formats)

1344-404: Is always useful or correct. For example, if an item complies with a certain standard, there is not necessarily assurance that it is fit for any particular use. The people who use the item or service (engineers, trade unions, etc.) or specify it (building codes, government, industry, etc.) have the responsibility to consider the available standards, specify the correct one, enforce compliance, and use

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1408-538: Is assigned a unique multicast address (in the range from 239.0.0.0 to 239.255.255.255); only one device can send to this address (many-to-many connections are not supported). To monitor keepalive status and allocate bandwidth, devices may use RTCP report interval, SIP session timers and OPTIONS ping, or ICMP Echo request (ping). AES67 uses DiffServ to set QoS traffic priorities in the Differentiated Services Code Point (DSCP) field of

1472-403: Is associated with each port. TC can belong to multiple clock domains and profiles. These provisions make it possible to synchronize IEEE 802.1AS clocks to IEEE 1588-2008 clocks used by AES67. The standard was developed by the Audio Engineering Society beginning at the end of 2010. The standard was initially published September 2013. A second printing which added a patent statement from Audinate

1536-455: Is defined for multicast connections. AES67 uses IEEE 1588-2008 Precision Time Protocol (PTPv2) for clock synchronisation. For standard networking equipment, AES67 defines configuration parameters for a "PTP profile for media applications", based on IEEE 1588 delay request-response sync and (optionally) peer-to-peer sync (IEEE 1588 Annexes J.3 and J4); event messages are encapsulated in IPv4 packets over UDP transport (IEEE 1588 Annex D). Some of

1600-419: Is defined, though it is recommended that devices shall adapt to packet time changes and/or determine packet time by analyzing RTP timestamps. Packet time determines RTP payload size according to a supported sample rate. 1 ms is required for all devices. Devices should support a minimum of 1 to 8 channels per stream. Network latency ( link offset ) is the time difference between the moment an audio stream enters

1664-519: Is recommended. Sources are required to maintain transmission with jitter of less than 17 packet times (or 17 ms if shorter), though 1 packet time (or 1 ms if shorter) is recommended. AES67 may transport media streams as IEEE 802.1BA AVB time-sensitive traffic Classes A and B on supported networks, with guaranteed latency of 2 ms and 50 ms respectively. Reservation of bandwidth with the Stream Reservation Protocol (SRP) specifies

1728-566: Is supported by RAVENNA-enabled devices under its AES67 Operational Profile. Over time this table will grow to become a resource for integration and compatibility between devices. The discovery methods supported by each device are critical for integration since the AES67 specification does not stipulate how this should be done, but instead provides a variety of options or suggestions. Also, AES67 specifies multicast and unicast but many AES67 devices only support multicast. Technical standard It

1792-452: Is the standard form of digital audio in computers, compact discs , digital telephony and other digital audio applications. In a PCM stream , the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves , Claude Shannon , Barney Oliver and John R. Pierce are credited with its invention. Linear pulse-code modulation ( LPCM )

1856-430: Is used over UDP transport. RTP payload is limited to 1460 bytes, to prevent fragmentation with default Ethernet MTU of 1500 bytes (after subtracting IP/UDP/RTP overhead of 20+8+12=40 Bytes). Contributing source (CSRC) identifiers and TLS encryption are not supported. Time synchronization, media stream delivery, and discovery protocols may use IP multicasting with IGMPv 2 (optionally IGMPv3) negotiation. Each media stream

1920-646: Is usually a formal document that establishes uniform engineering or technical criteria, methods, processes, and practices. In contrast, a custom, convention, company product, corporate standard, and so forth that becomes generally accepted and dominant is often called a de facto standard. A technical standard may be developed privately or unilaterally, for example by a corporation, regulatory body, military, etc. Standards can also be developed by groups such as trade unions and trade associations. Standards organizations often have more diverse input and usually develop voluntary standards: these might become mandatory if adopted by

1984-460: The Internet protocol suite . AES67 also defines audio sample format and sample rate, supported number of channels, as well as IP data packet size and latency/buffering requirements. The standard calls out several protocol options for device discovery but does not require any to be implemented. Session Initiation Protocol is used for unicast connection management. No connection management protocol

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2048-652: The SIGSALY encryption equipment, conveyed high-level Allied communications during World War II . In 1943 the Bell Labs researchers who designed the SIGSALY system became aware of the use of PCM binary coding as already proposed by Reeves. In 1949, for the Canadian Navy's DATAR system, Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances. PCM in

2112-583: The WTO does not rule out the possibility that the actions of private standard-setting bodies may be subject to WTO law. BSI Group compared private food safety standards with "plugs and sockets", explaining the food sector is full of "confusion and complexity". Also, "the multiplicity of standards and assurance schemes has created a fragmented and inefficient supply chain structure imposing unnecessary costs on businesses that have no choice but to pass on to consumers". BSI provide examples of other sectors working with

2176-474: The public switched telephone network (PSTN) had been largely digitized with very-large-scale integration (VLSI) CMOS PCM codec-filters, widely used in electronic switching systems for telephone exchanges , user-end modems and a wide range of digital transmission applications such as the integrated services digital network (ISDN), cordless telephones and cell phones . PCM is the method of encoding typically used for uncompressed digital audio. In

2240-453: The AES published a report describing synchronization interoperability between AES67 and SMPTE 2059-2 . In June 2016, AES67 audio transport enhanced by AVB/TSN clock synchronisation and bandwidth reservation was demonstrated at InfoComm 2016. In September 2017, SMPTE published ST 2110, a standard for professional video over IP . ST 2110-30 uses AES67 as the transport for audio accompanying

2304-468: The Endorsement of Forest Certification (PEFC) issued position statements defending their use of private standards in response to reports from The Institute for Multi-Stakeholder Initiative Integrity (MSI Integrity) and Greenpeace. Private standards typically require a financial contribution in terms of an annual fee from the organizations who adopt the standard. Corporations are encouraged to join

2368-628: The Expedited Forwarding (EF) class, which typically implements strict priority per-hop behavior (PHB). All other traffic is handled on a best effort basis with Default Forwarding. RTP Clock Source Signalling procedure is used to specify PTP domain and grandmaster ID for each media stream. Sample formats include 16-bit and 24-bit Linear PCM with 48 kHz sampling frequency, and optional 24-bit 96 kHz and 16-bit 44.1 kHz. Other RTP audio video formats may be supported. Multiple sample frequencies are optional. Devices may enforce

2432-524: The IP packet. Three classes should be supported at a minimum: 250 μs maximum delay may be required for time-critical applications to prevent drops of audio. To prioritize critical media streams in a large network, applications may use additional values in the Assured Forwarding class 4 with low-drop probability (AF41), typically implemented as a weighted round-robin queue. Clock traffic is assigned to

2496-408: The amount of traffic generated through a measurement interval of 125 μs and 250 μs respectively. Multicast IP addresses have to be used, though only with a single source, as AVB networks only support Ethernet multicast destination addressing in the range from 01:00:5e:00:00:00 to 01:00:5e:7f:ff:ff. An SRP talker advertise message shall be mapped as follows: Under both IEEE 1588-2008 and IEEE 802.1AS,

2560-439: The benefits have been debated. The Nyquist–Shannon sampling theorem shows PCM devices can operate without introducing distortions within their designed frequency bands if they provide a sampling frequency at least twice that of the highest frequency contained in the input signal. For example, in telephony , the usable voice frequency band ranges from approximately 300  Hz to 3400 Hz. For effective reconstruction of

2624-443: The board of governance of the standard owner which enables reciprocity. Meaning corporations have permission to exert influence over the requirements in the standard, and in return the same corporations promote the standards in their supply chains which generates revenue and profit for the standard owner. Financial incentives with private standards can result in a perverse incentive , where some private standards are created solely with

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2688-445: The channel. In other cases, extra framing bits are added into the stream, which guarantees at least occasional symbol transitions. Another technique used to control ones-density is the use of a scrambler on the data, which will tend to turn the data stream into a stream that looks pseudo-random , but where the data can be recovered exactly by a complementary descrambler. In this case, long runs of zeroes or ones are still possible on

2752-512: The default parameters are adjusted, specifically, logSyncInterval and logMinDelayReqInterval are reduced to improve accuracy and startup time. Clock Grade 2 as defined in AES11 Digital Audio Reference Signal (DARS) is signaled with clockClass. Network equipment conforming to IEEE 1588-2008 uses default PTP profiles; for video streams, SMPTE 2059-2 PTP profile can be used. In AVB/TSN networks, synchronization

2816-461: The development of a technical standard, private standards adopt a non-consensus process in comparison to voluntary consensus standards. This is explained in the paper International standards and private standards . The International Trade Centre published a literature review series with technical papers on the impacts of private standards and the Food and Agriculture Organization (FAO) published

2880-539: The diagram, a sine wave (red curve) is sampled and quantized for PCM. The sine wave is sampled at regular intervals, shown as vertical lines. For each sample, one of the available values (on the y-axis) is chosen. The PCM process is commonly implemented on a single integrated circuit called an analog-to-digital converter (ADC). This produces a fully discrete representation of the input signal (blue points) that can be easily encoded as digital data for storage or manipulation. Several PCM streams could also be multiplexed into

2944-419: The digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques, such as modified discrete cosine transform (MDCT) coding. In telephony, a standard audio signal for a single phone call is encoded as 8,000 samples per second , of 8 bits each, giving a 64 kbit/s digital signal known as DS0 . The default signal compression encoding on

3008-556: The first commercial digital recordings. In 1972, Denon unveiled the first 8-channel digital recorder, the DN-023R, which used a 4-head open reel broadcast video tape recorder to record in 47.25 kHz, 13-bit PCM audio. In 1977, Denon developed the portable PCM recording system, the DN-034R. Like the DN-023R, it recorded 8 channels at 47.25 kHz, but it used 14-bits "with emphasis , making it equivalent to 15.5 bits." In 1979,

3072-437: The first digital pop album, Bop till You Drop , was recorded. It was recorded in 50 kHz, 16-bit linear PCM using a 3M digital tape recorder. The compact disc (CD) brought PCM to consumer audio applications with its introduction in 1982. The CD uses a 44,100 Hz sampling frequency and 16-bit resolution and stores up to 80 minutes of stereo audio per disc. The rapid development and wide adoption of PCM digital telephony

3136-506: The intent of generating money. BRCGS, as scheme owner of private standards, was acquired in 2016 by LGC Ltd who were owned by private equity company Kohlberg Kravis Roberts . This acquisition triggered substantial increases in BRCGS annual fees. In 2019, LGC Ltd was sold to private equity companies Cinven and Astorg. Linear PCM Pulse-code modulation ( PCM ) is a method used to digitally represent analog signals . It

3200-411: The item correctly. Validation of suitability is necessary. Standards often get reviewed, revised and updated on a regular basis. It is critical that the most current version of a published standard be used or referenced. The originator or standard writing body often has the current versions listed on its web site. In social sciences , including economics , a standard is useful if it is a solution to

3264-449: The late 1940s and early 1950s used a cathode-ray coding tube with a plate electrode having encoding perforations. As in an oscilloscope , the beam was swept horizontally at the sample rate while the vertical deflection was controlled by the input analog signal, causing the beam to pass through higher or lower portions of the perforated plate. The plate collected or passed the beam, producing current variations in binary code, one bit at

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3328-769: The original analog signal: the sampling rate , which is the number of times per second that samples are taken; and the bit depth , which determines the number of possible digital values that can be used to represent each sample. Early electrical communications started to sample signals in order to multiplex samples from multiple telegraphy sources and to convey them over a single telegraph cable. The American inventor Moses G. Farmer conceived telegraph time-division multiplexing (TDM) as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanical commutator for time-division multiplexing multiple telegraph signals; he also applied this technology to telephony . He obtained intelligible speech from channels sampled at

3392-521: The output but are considered unlikely enough to allow reliable synchronization. In other cases, the long term DC value of the modulated signal is important, as building up a DC bias will tend to move communications circuits out of their operating range. In this case, special measures are taken to keep a count of the cumulative DC bias and to modify the codes if necessary to make the DC bias always tend back to zero. Many of these codes are bipolar codes , where

3456-403: The packet time. Small buffers decrease latency but may result in drops of audio when media data does not arrive on time. Unexpected changes to network conditions and jitter from packet encoding and processing may require longer buffering and therefore higher latency. Destinations are required to use a buffer of 3 times the packet time, though at least 20 times the packet time (or 20 ms if smaller)

3520-540: The pulses can be positive, negative or absent. In the typical alternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes. The word pulse in the term pulse-code modulation refers to the pulses to be found in the transmission line. This perhaps is a natural consequence of this technique having evolved alongside two analog methods, pulse-width modulation and pulse-position modulation , in which

3584-514: The same title was filed by John R. Pierce in 1945, and issued in 1948: U.S. patent 2,437,707 . The three of them published "The Philosophy of PCM" in 1948. The T-carrier system, introduced in 1961, uses two twisted-pair transmission lines to carry 24 PCM telephone calls sampled at 8 kHz and 8-bit resolution. This development improved capacity and call quality compared to the previous frequency-division multiplexing schemes. In 1973, adaptive differential pulse-code modulation (ADPCM)

3648-476: The source (ingress time), marked by RTP timestamp in the media packet, and the moment it leaves the destination (egress time). Latency depends on packet time, propagation and queuing delays, packet processing overhead, and buffering in the destination device; thus minimum latency is the shortest packet time and network forwarding time, which can be less than 1 μs on a point-to-point Gigabit Ethernet link with minimum packet size, but in real-world networks could be twice

3712-585: The use of PCM for voice communication in 1937 while working for International Telephone and Telegraph in France. He described the theory and its advantages, but no practical application resulted. Reeves filed for a French patent in 1938, and his US patent was granted in 1943. By this time Reeves had started working at the Telecommunications Research Establishment . The first transmission of speech by digital techniques,

3776-620: The video. In December 2017 the Media Networking Alliance merged with the Alliance for IP Media Solutions (AIMS) combining efforts to promote standards-based network transport for audio and video. In April 2018 AES67-2018 was published. The principal change in this revision is addition of a protocol implementation conformance statement (PICS). The AES Standards Committee and AES67 editor, Kevin Gross, were recipients of

3840-692: The voice signal even further. An ADPCM algorithm is used to map a series of 8-bit μ-law or A-law PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard. Audio coding formats and audio codecs have been developed to achieve further compression. Some of these techniques have been standardized and patented. Advanced compression techniques, such as modified discrete cosine transform (MDCT) and linear predictive coding (LPC), are now widely used in mobile phones , voice over IP (VoIP) and streaming media . PCM can be either return-to-zero (RZ) or non-return-to-zero (NRZ). For

3904-452: The voice signal, telephony applications therefore typically use an 8000 Hz sampling frequency which is more than twice the highest usable voice frequency. Regardless, there are potential sources of impairment implicit in any PCM system: Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the analog-to-digital process; newer implementations do so in

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3968-564: Was developed, by P. Cummiskey, Nikil Jayant and James L. Flanagan . In 1967, the first PCM recorder was developed by NHK 's research facilities in Japan. The 30 kHz 12-bit device used a compander (similar to DBX Noise Reduction ) to extend the dynamic range, and stored the signals on a video tape recorder . In 1969, NHK expanded the system's capabilities to 2-channel stereo and 32 kHz 13-bit resolution. In January 1971, using NHK's PCM recording system, engineers at Denon recorded

4032-438: Was enabled by metal–oxide–semiconductor (MOS) switched capacitor (SC) circuit technology, developed in the early 1970s. This led to the development of PCM codec-filter chips in the late 1970s. The silicon-gate CMOS (complementary MOS) PCM codec-filter chip, developed by David A. Hodges and W.C. Black in 1980, has since been the industry standard for digital telephony. By the 1990s, telecommunication networks such as

4096-619: Was published in March 2014. The Media Networking Alliance was formed in October 2014 to promote adoption of AES67. In October 2014 a plugfest was held to test interoperability achieved with AES67. A second plugfest was conducted in November 2015 and third in February 2017. An update to the standard including clarifications and error corrections was issued in September 2015. In May 2016,

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