CN112601077B - Automatic encoder delay measuring method based on audio - Google Patents
Automatic encoder delay measuring method based on audio Download PDFInfo
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- CN112601077B CN112601077B CN202011439857.5A CN202011439857A CN112601077B CN 112601077 B CN112601077 B CN 112601077B CN 202011439857 A CN202011439857 A CN 202011439857A CN 112601077 B CN112601077 B CN 112601077B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N17/00—Diagnosis, testing or measuring for television systems or their details
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N19/00—Methods or arrangements for coding, decoding, compressing or decompressing digital video signals
- H04N19/40—Methods or arrangements for coding, decoding, compressing or decompressing digital video signals using video transcoding, i.e. partial or full decoding of a coded input stream followed by re-encoding of the decoded output stream
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
- H04N21/21—Server components or server architectures
- H04N21/218—Source of audio or video content, e.g. local disk arrays
- H04N21/2187—Live feed
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/80—Generation or processing of content or additional data by content creator independently of the distribution process; Content per se
- H04N21/85—Assembly of content; Generation of multimedia applications
- H04N21/854—Content authoring
- H04N21/8547—Content authoring involving timestamps for synchronizing content
Abstract
The invention discloses an automatic measuring method of encoder delay based on audio. The method specifically comprises the following steps: (1) generating a section of material frame by using an audio time coding mode, compressing and packaging the material into a CBR mode TS file by using an AAC coding mode, starting audio time coding of the material from 0, converting an audio PTS (partial program instruction) from 0 into an audio time coding time stamp through the PTS, and sending the test material out by using UDP (user datagram protocol) through a test net port according to a set CBR code rate to generate a test stream; (2) configuring a tested transcoder to receive the test stream, and sending the test stream back to the test equipment after transcoding; (3) and the test equipment receives the returned test stream, decodes, detects and extracts the timestamp, and compares the timestamp with the source test stream to obtain integral delay data. The invention has the beneficial effects that: the time delay is detected by detecting the time coding on the audio, the method has universality, is not interfered by a player and a transmission layer, and can track the time delay jitter in real time.
Description
Technical Field
The invention relates to the technical field of coders, in particular to an automatic encoder delay measuring method based on audio.
Background
In the field of live video, delay is an important factor affecting the viewer experience, so that delay measurement and evaluation of a transcoding system are necessary. The end-to-end delay of the transcoder refers to the delay of the whole process of receiving, de-encapsulating, decoding, preprocessing, encoding, encapsulating and sending the media code stream. The methods generally used are: the stream pushing end and the playing end are opposite to the same clock, and then the time displayed by the stream pushing end is subtracted from the time displayed by the playing end, so that the live broadcast delay is obtained. The disadvantages of this approach are many, 1) imprecision, 2) player-influenced, 3) transport-layer-influenced, 4) manual involvement, and 5) delay jitter cannot be tracked in real-time.
Disclosure of Invention
The invention provides an automatic measuring method of encoder delay based on audio frequency, which can track delay jitter in real time.
In order to achieve the purpose, the invention adopts the following technical scheme:
an automatic measuring method of encoder time delay based on audio frequency specifically comprises the following steps:
(1) generating a section of material frame by using an audio time coding mode, compressing and packaging the material into a CBR mode TS file by using an AAC coding mode, starting audio time coding of the material from 0, converting an audio PTS (partial program instruction) from 0 into an audio time coding time stamp through the PTS, and sending the test material out by using UDP (user datagram protocol) through a test net port according to a set CBR code rate to generate a test stream;
(2) configuring a tested transcoder to receive the test stream, and sending the test stream back to the test equipment after transcoding;
(3) and the test equipment receives the returned test stream, decodes, detects and extracts the timestamp, and compares the timestamp with the source test stream to obtain integral delay data.
The audio needs to be identified when the delay is measured, the identification method can be performed in a mode similar to a custom private timestamp in an SEI (media independent interface), but the custom information is generally discarded after transcoding by transcoding equipment, so that the measurement cannot be performed. The invention uses an audio time coding mode to mark time, detects time delay by detecting time coding on audio, and has universality. The invention provides an automatic measurement technology based on the whole delay of a coding and transcoding device aiming at TS CBR code stream transmitted by UDP (transport stream buffer) in the field of broadcasting and television, is not interfered by a player and a transmission layer, and can track delay jitter in real time.
Preferably, in step (1), the audio time coding scheme is specifically as follows: the audio waveform is encoded by using a square wave or positive selection wave with 440Hz and certain amplitude at a mute interval of 1 second, the duration of the first section is 1 second, the duration of the second section is 2 seconds, and so on, the duration of the last section is N seconds, and N is a preset value.
Preferably, the extraction of the audio temporal coding is specifically as follows: judging the mute interval by scanning the duration of the 0 value, firstly judging whether the interval is the continuous 0 value, if so, neglecting the current 0 value duration, positioning a first non-0 section, scanning backwards from the first non-0 section, extracting a first mute section, judging the number of sampling points within 1 second according to the mute section, and judging the time of a certain sampling point according to the number of the sampling points within 1 second.
Preferably, in step (1), the sending flow of the test flow specifically includes: setting a detected test material file and an udp sending address, analyzing the detected test material file, calculating a code rate according to a PCR of the test material file, relocating to the beginning of the test material file to start sending, after reading a data packet, analyzing the PCR if the PCR exists, analyzing the PTS of audio if the PTS exists, calculating an audio time coding timestamp according to the PTS, wherein the audio time coding of the test material starts from 0, the audio PTS also starts from 0, sending the data packet and obtaining a sending timestamp, and recording a pair of the sending timestamp and the audio time coding timestamp into an array.
Preferably, in step (1), the sending time stamp is obtained from the network card through the socket interface, the time stamp is the time when the network card sends the audio to the network, if the PTS of the first frame audio of the test material file does not start from 0, the offset is recorded, and the audio time coding time stamp is corrected by subtracting the offset from the rest PTS.
Preferably, in step (2), the receiving procedure of the test flow specifically includes: setting a udp receiving address, receiving an udp data packet, extracting a receiving timestamp of the udp data packet, analyzing a PTS of an audio if the PTS exists, recording a pair of the receiving timestamp and the PTS into an array, detecting an audio time coding timestamp after decoding, inquiring data of the pair of the receiving timestamp and the PTS, finding a corresponding receiving timestamp according to the PTS, replacing the PTS to update the audio time coding timestamp, and obtaining a pair of the receiving timestamp and the audio time coding timestamp as a final result.
Preferably, in step (2), the receiving timestamp is obtained from the network card through the socket interface, the timestamp is the time when the network card sends the packet to the network, and the timestamp adopts a circular buffering mechanism for the packet.
Preferably, in step (3), the detection process of the test equipment specifically includes: after receiving a new pair of array of the receiving timestamp and the audio time coding timestamp, the pair of array of the transmitting timestamp and the audio time coding timestamp is searched according to the audio time coding timestamp, the pair of array of the receiving timestamp and the audio time coding timestamp is searched according to the audio time coding timestamp, and the latest delay time is displayed as the receiving timestamp-the transmitting timestamp.
The beneficial effects of the invention are: the time delay is detected by detecting the time coding on the audio, the method has universality, is not interfered by a player and a transmission layer, and can track the time delay jitter in real time.
Drawings
FIG. 1 is a schematic diagram of an audio time coding mode square wave of the present invention;
FIG. 2 is a schematic diagram of an audio time-coding mode sine wave according to the present invention;
FIG. 3 is a schematic diagram of an audio time encoding method according to the present invention;
FIG. 4 is a schematic diagram of audio temporal coding extraction according to the present invention;
FIG. 5 is a schematic diagram illustrating the flow of test flow transmission;
FIG. 6 is a schematic diagram of the flow of receiving the test flow of the present invention;
FIG. 7 is a schematic diagram of the detection process of the test apparatus of the present invention.
Detailed Description
The invention is further described with reference to the following figures and detailed description.
An automatic measuring method of encoder time delay based on audio frequency specifically comprises the following steps:
(1) generating a section of material frame by an audio time coding mode, preferably having a length of dozens of seconds, compressing and packaging the material into a CBR mode TS file by using an AAC coding mode, for simplicity, starting audio time coding of the material from 0, starting audio PTS from 0, converting the material into an audio time coding time stamp through PTS, and sending the test material out by UDP according to a set CBR code rate through a test net port to generate a test stream;
as shown in fig. 3, the audio time coding method is specifically as follows:
the audio temporal coding scheme is specifically as follows: the audio waveform is encoded by using a square wave or positive selection wave (as shown in fig. 1 and fig. 2) with 440Hz and a certain amplitude at mute intervals of 1 second, wherein the duration of the first segment is 1 second, the duration of the second segment is 2 seconds, and so on, the duration of the last segment is N seconds, N is a preset value, and N is generally 20 seconds, which can be properly prolonged.
As shown in fig. 4, the extraction of audio temporal coding is specifically as follows: because the encoder can perform audio resampling, the judgment cannot be performed through the number of original sampling points, the mute interval is judged by scanning the duration (1 second) of a 0 value, a square wave has no 0 value in a non-mute period, a sine wave has a 0 value in the non-mute period, the scanning process is performed by paying attention to neglect the 0 value with short duration, firstly, whether the square wave is the continuous 0 value is judged, if the square wave is the continuous 0 value, the current 0 value duration is ignored, the first non-0 section is positioned, the backward scanning is started from the first non-0 section, the first mute section is extracted, the number of sampling points in 1 second is judged according to the mute section, and the time of a certain sampling point is judged according to the number of sampling points in 1 second.
As shown in fig. 5, the sending process of the test flow specifically includes: setting a detected test material file and an udp sending address, analyzing the detected test material file, calculating a code rate according to a PCR of the test material file, relocating to the beginning of the test material file to start sending, reading a data packet, analyzing the PCR if the PCR exists, analyzing the PTS of an audio if the PTS exists, calculating an audio time coding time stamp according to the PTS, wherein the audio time coding of the test material starts from 0, the audio PTS also starts from 0, sending the data packet and obtaining a sending time stamp, recording a pair of the sending time stamp and the audio time coding time stamp into an array, and considering that the time delay of a transcoder is generally several seconds, the data size can be limited to 4K (about 8 seconds for 60fps materials), and the data size can be increased properly. The sending time stamp is obtained from the network card through the socket interface (the network card needs to support), and the time stamp is the time when the network card sends to the network, so that the jitter of the kernel and the protocol stack is avoided. If the PTS of the first frame of audio of the test material file does not start from 0, this offset is recorded, and the remaining PTS's are subtracted by the offset to correct the audio time coding time stamp.
(2) Configuring a tested transcoder to receive the test stream, and sending the test stream back to the test equipment after transcoding;
as shown in fig. 6, the receiving process of the test flow specifically includes: setting a udp receiving address, receiving an udp data packet, extracting a receiving timestamp of the udp data packet, analyzing a PTS of an audio if the PTS exists, recording a pair of the receiving timestamp and the PTS into an array, detecting an audio time coding timestamp after decoding, inquiring data of the pair of the receiving timestamp and the PTS, finding a corresponding receiving timestamp according to the PTS, replacing the PTS to update the audio time coding timestamp, and obtaining a pair of the receiving timestamp and the audio time coding timestamp as a final result. And a receiving time stamp is acquired from the network card through the socket interface (the network card needs to support), and the time stamp is the time when the network card sends the network card to the network, so that the jitter of a kernel and a protocol stack is avoided. The timestamp adopts a circular buffer mechanism for the array, and memory copy is avoided. Because the data volume is large, the array search adopts a quick search algorithm such as binary search.
(3) The test equipment receives the returned test stream, decodes, detects and extracts the timestamp, and compares the timestamp with the source test stream to obtain integral delay data;
as shown in fig. 7, the detection process of the test equipment specifically includes: after receiving the new pair of array of the receiving time stamp and the audio time coding time stamp, searching the pair of array of the sending time stamp and the audio time coding time stamp according to the audio time coding time stamp, searching the pair of array of the receiving time stamp and the audio time coding time stamp according to the audio time coding time stamp, and displaying the latest delay time as the receiving time stamp-the sending time stamp.
Wherein: the test scheme can directly connect the whole time delay of the tested transcoder with the tested equipment, and can also measure the whole time delay of the transmission network and the transcoder through the transmission network, and trace and display the time delay jitter curve. The embodiment adopts the network port transmission network mode to measure the delay. From the project implementation effect, the delay data can be automatically analyzed and is accurate to the frame level.
The audio needs to be identified when the delay is measured, the identification method can be performed in a mode similar to a custom private timestamp in an SEI (media independent interface), but the custom information is generally discarded after transcoding by transcoding equipment, so that the measurement cannot be performed. The invention uses an audio time coding mode to mark time, detects time delay by detecting time coding on audio, and has universality. The invention provides an automatic measurement technology based on the whole delay of a coding and transcoding device aiming at TS CBR code stream transmitted by UDP (transport stream buffer) in the field of broadcasting and television, is not interfered by a player and a transmission layer, and can track delay jitter in real time.
Claims (5)
1. An automatic measuring method of encoder time delay based on audio frequency is characterized by comprising the following steps:
(1) generating a section of material frame by using an audio time coding mode, compressing and packaging the material into a CBR mode TS file by using an AAC coding mode, starting audio time coding of the material from 0, starting audio PTS from 0, converting the audio time coding time stamp into an audio time coding time stamp through PTS, sending out the test material by UDP according to a set CBR code rate through a test network port, and generating a test stream; the audio temporal coding scheme is specifically as follows: using 440Hz square waves or positive selection waves with certain amplitude to encode audio waveforms at 1-second mute intervals, wherein the duration of a first section is 1 second, the duration of a second section is 2 seconds, and so on, the duration of the last section is N seconds, and N is a preset value; the extraction of the audio temporal coding is specifically as follows: judging a mute interval by scanning the duration time of a 0 value, firstly judging whether the interval is a continuous 0 value, if so, neglecting the current 0 value duration section, positioning a first non-0 section, scanning backwards from the first non-0 section, extracting a first mute section, judging the number of sampling points within 1 second according to the mute section, and judging the time of a certain sampling point according to the number of the sampling points within 1 second;
(2) configuring a tested transcoder to receive the test stream, and sending the test stream back to the test equipment after transcoding;
(3) the test equipment receives the returned test stream, decodes, detects and extracts the timestamp, and compares the timestamp with the source test stream to obtain integral delay data; the detection process of the test equipment specifically comprises the following steps: after receiving the new pair of array of the receiving time stamp and the audio time coding time stamp, the pair of array of the sending time stamp and the audio time coding time stamp is searched according to the audio time coding time stamp, the pair of array of the receiving time stamp and the audio time coding time stamp is searched according to the audio time coding time stamp, and the delay time = the receiving time stamp-the sending time stamp, and the latest delay time is displayed.
2. The method according to claim 1, wherein in step (1), the test stream is sent by: setting a test material file and an udp sending address for detection, analyzing the detected test material file, calculating a code rate according to a PCR of the test material file, repositioning the code rate to the beginning of the test material file to start sending, reading a data packet, analyzing the PCR if the PCR exists, analyzing the PTS of audio if the PTS exists, calculating an audio time coding timestamp according to the PTS, sending the data packet and acquiring a sending timestamp, and recording the pair of the sending timestamp and the audio time coding timestamp into an array.
3. The method of claim 1, wherein in step (1), the transmission time stamp is obtained from the network card through the socket interface, the time stamp is the time when the network card transmits to the network, if the PTS of the first frame of audio of the test material file is not from 0, the offset is recorded, and the audio time coding time stamp is corrected by subtracting the offset from the remaining PTS.
4. The method according to claim 2, wherein in the step (2), the test stream is received by: setting a udp receiving address, receiving an udp data packet, extracting a receiving timestamp of the udp data packet, analyzing a PTS of an audio if the PTS exists, recording a pair of the receiving timestamp and the PTS into an array, detecting an audio time coding timestamp after decoding, inquiring data of the pair of the receiving timestamp and the PTS, finding a corresponding receiving timestamp according to the PTS, replacing the PTS to update the audio time coding timestamp, and obtaining a pair of the receiving timestamp and the audio time coding timestamp as a final result.
5. The method as claimed in claim 4, wherein in step (2), the receiving timestamp is obtained from the network card through the socket interface, the timestamp is the time when the network card sends the timestamp to the network, and the timestamp employs a circular buffering mechanism for the groups.
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