CN100488133C - Method for detecting multimedia terminal audio frequency service quality performance - Google Patents
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Abstract
Description
技术领域 technical field
本发明涉及一种多媒体终端音频服务质量特性的测试方法。The invention relates to a testing method for audio service quality characteristics of a multimedia terminal.
背景技术 Background technique
在多媒体终端的开发和测试中,常常需要测定多媒体终端音频的QoS(Quality of Service,服务质量)特性,以衡量不同网络状态下,多媒体终端的服务带给终端用户的实际感受并获得终端内部QoS算法的重要特性。当前,对于多媒体终端音频的QoS特性的测量包括与网络或者播放缓冲相关的多媒体终端音频的QoS特性,主要是基于最小化网络延迟和网络丢包的权衡,从各个环节提供改善终端QoS性能的算法,包括码流封装的改动,对网络延迟的估计以及对播放缓冲区的自适应调整等,这些都偏重于理论的推导和分析。因而,对于实际测量而言,这种推导分析得到的结果准确度较低,且使用起来较为复杂。In the development and testing of multimedia terminals, it is often necessary to measure the QoS (Quality of Service) characteristics of multimedia terminal audio to measure the actual experience of multimedia terminal services to end users under different network conditions and to obtain terminal internal QoS Important properties of algorithms. At present, the measurement of the QoS characteristics of multimedia terminal audio includes the QoS characteristics of multimedia terminal audio related to the network or playback buffer, mainly based on the trade-off between minimizing network delay and network packet loss, and providing algorithms to improve terminal QoS performance from various links , including the change of code stream encapsulation, the estimation of network delay and the adaptive adjustment of playback buffer, etc., all of which focus on theoretical derivation and analysis. Therefore, for actual measurement, the accuracy of the results obtained by such derivation and analysis is low, and it is more complicated to use.
发明内容 Contents of the invention
本发明要解决的技术问题是提供一种可较为准确获得多媒体终端内部音频服务质量特性的多媒体终端音频服务质量特性的测试方法。The technical problem to be solved by the present invention is to provide a testing method for the audio quality of service characteristics of the multimedia terminal which can obtain the audio quality of service characteristics inside the multimedia terminal more accurately.
本发明解决其技术问题所采用的技术方案是:The technical solution adopted by the present invention to solve its technical problems is:
一种多媒体终端音频服务质量特性的测试方法,包括以下步骤:A method for testing audio quality of service characteristics of a multimedia terminal, comprising the following steps:
1.1 在模拟终端音频发送部分加入用于模拟实际网络的音频服务质量特性测试模块,并将音频经过所述模拟终端音频服务质量特性测试模块得到的测试数据集分成两路,一路进入音宿计算机,另一路由模拟终端通过网络发送给待测的所述多媒体终端后进入所述音宿计算机,最后使所述模拟终端与所述待测的多媒体终端通过所述网络交互;1.1 Add an audio quality of service characteristic test module for simulating the actual network in the audio transmission part of the analog terminal, and divide the test data set obtained by passing the audio quality of service characteristic test module of the analog terminal audio into two paths, and one path enters the audio computer, Another routing simulation terminal is sent to the multimedia terminal to be tested through the network and then enters the audio sink computer, and finally the simulation terminal is interacted with the multimedia terminal to be tested through the network;
1.2 当前音频经过所述模拟终端后形成当前测试数据集,所述音宿计算机分别记录所述当前测试数据集进入所述音宿计算机的第一时间与所述当前测试数据集经过所述多媒体终端后进入所述音宿计算机的第二时间;1.2 After the current audio passes through the analog terminal, the current test data set is formed, and the audio sink computer records the first time when the current test data set enters the audio sink computer and the current test data set passes through the multimedia terminal. After entering the second time of the sound sink computer;
1.3 根据所述第一时间与所述第二时间判断所述多媒体终端音频的服务质量特性,若要继续测试,则回到步骤1.2,否则结束测试。1.3 Judging the quality of service characteristics of the multimedia terminal audio according to the first time and the second time, if the test is to be continued, return to step 1.2, otherwise end the test.
上述方案中,所述测试数据集为突发延迟测试数据集。In the above solution, the test data set is a burst delay test data set.
上述方案中,所述测试数据集为振荡延迟测试数据集。In the above solution, the test data set is an oscillation delay test data set.
上述方案中,所述测试数据集为延迟跳变测试数据集。In the above solution, the test data set is a delay hopping test data set.
上述方案中,所述测试数据集为步进延迟测试数据集。In the above solution, the test data set is a step delay test data set.
上述方案中,所述音频服务质量特性测试模块包括延迟、抖动和丢包模块。In the above solution, the audio quality of service characteristic testing module includes delay, jitter and packet loss modules.
上述方案中,所述音宿计算机对所述第一时间和所述第二时间的记录是通过记录所述测试数据集的实时传送协议包的时间实现的。In the above solution, the record of the first time and the second time by the sound sink computer is realized by recording the time of the real-time transport protocol packet of the test data set.
上述方案中,所述模拟终端与所述待测的多媒体终端接在同一个集线器或者交换机上。In the above solution, the analog terminal and the multimedia terminal to be tested are connected to the same hub or switch.
上述方案中,所述音频由音源计算机或者所述模拟终端输出。In the solution above, the audio is output by the audio source computer or the analog terminal.
上述方案中,所述音频为基本不消耗处理能力的格式。In the above solution, the audio is in a format that basically does not consume processing power.
本发明的有益效果主要表现在:本发明提供的技术方案结合适当的测试数据集,可以较为准确获得多媒体终端内部的音频QoS特性,并可大致推断出终端的QoS算法。The beneficial effect of the present invention is mainly manifested in that: the technical solution provided by the present invention combined with an appropriate test data set can obtain the audio QoS characteristics inside the multimedia terminal more accurately, and roughly deduce the QoS algorithm of the terminal.
附图说明 Description of drawings
图1为本发明一实施例的测试部署示意图;FIG. 1 is a schematic diagram of test deployment according to an embodiment of the present invention;
图2为本发明一实施例的突发延迟测试数据集特性图;Fig. 2 is a characteristic diagram of a burst delay test data set according to an embodiment of the present invention;
图3为本发明一实施例的振荡延迟测试数据集特性图;3 is a characteristic diagram of an oscillation delay test data set according to an embodiment of the present invention;
图4为本发明一实施例利用延迟跳变测试数据集测定播放缓冲最大能力的示意图;Fig. 4 is a schematic diagram of determining the maximum capability of playback buffering using a delay jump test data set according to an embodiment of the present invention;
图5为本发明一实施例的步进延迟测试数据集特性图。FIG. 5 is a characteristic diagram of a step delay test data set according to an embodiment of the present invention.
具体实施方式 Detailed ways
下面结合附图对本发明作进一步的描述。The present invention will be further described below in conjunction with the accompanying drawings.
参照图1,一种多媒体终端音频服务质量特性的测试方法,首先在模拟终端音频发送部分加入音频服务质量特性测试模块,其包括延迟、抖动和丢包模块,以模拟实际网络可能遇到的情况;并将音频经过模拟终端音频服务质量特性测试模块得到的测试数据集分成两路,一路直接进入音宿计算机,另一路由模拟终端通过网络发送给待测的多媒体终端后进入音宿计算机。这样,在音宿计算机上采集的两路音频的延迟就包含了待测的多媒体终端对音频的处理延迟。其中,音频为在音源计算机或者模拟终端上准备好的音频素材,其被预处理成G711等基本不消耗处理能力的格式;模拟终端和音源计算机可为同一物理计算机,而模拟终端与待测的多媒体终端则接在同一个集线器或者交换机上以最小化网络延迟对测试结果的影响。完成测试部署后,使模拟终端与待测的多媒体终端通过所述网络交互并进入测试过程。With reference to Fig. 1, a kind of testing method of multimedia terminal audio quality of service characteristic, at first add the audio quality of service characteristic test module in the analog terminal audio transmission part, it includes delay, jitter and packet loss module, to simulate the situation that actual network may encounter and divide the test data set obtained by the analog terminal audio service quality characteristic test module into two paths, one path directly enters the audio sink computer, and the other route analog terminal sends to the multimedia terminal to be tested through the network and then enters the audio sink computer. In this way, the delay of the two channels of audio collected on the audio sink computer includes the audio processing delay of the multimedia terminal to be tested. Among them, the audio is the audio material prepared on the sound source computer or the analog terminal, which is preprocessed into a format such as G711 that basically does not consume processing power; the analog terminal and the sound source computer can be the same physical computer, and the analog terminal and the tested Multimedia terminals are connected to the same hub or switch to minimize the impact of network delay on test results. After the test deployment is completed, the simulated terminal interacts with the multimedia terminal to be tested through the network and enters the test process.
在测试过程中,当前音频经过所述模拟终端后形成当前测试数据集,音宿计算机通过记录RTP(Real-time Transport Protocol,实时传送协议)包的时间的方式分别记录当前测试数据集进入音宿计算机的第一时间与当前测试数据集经过多媒体终端后进入音宿计算机的第二时间;最后,音宿计算机根据所述第一时间与所述第二时间判断所述多媒体终端音频的服务质量特性。至此,本次测试结束,若要继续测试则将测试部署中各设备复位并重复上述测试过程。In the test process, the current audio frequency passes through the simulated terminal to form the current test data set, and the computer of the sound station records the time of the RTP (Real-time Transport Protocol, real-time transfer protocol) packet and enters the sound place respectively. The first time of the computer and the second time when the current test data set enters the audio sink computer after passing through the multimedia terminal; finally, the audio sink computer judges the quality of service characteristics of the multimedia terminal audio according to the first time and the second time . At this point, the test is over. If you want to continue the test, reset each device in the test deployment and repeat the above test process.
下面结合附图对音频通过模拟终端得到的测试数据集进行描述,图2、图3、图4中,横坐标为数据点,纵坐标为时间,2号曲线为数据点实际播放时间曲线,3号曲线为数据点理论播放时间曲线,1号曲线即为2号与3号曲线所表示的数据点播放延迟的变化曲线。The following is a description of the test data set obtained by audio through the analog terminal in conjunction with the accompanying drawings. In Figure 2, Figure 3, and Figure 4, the abscissa is the data point, the ordinate is time, and the No. 2 curve is the actual playback time curve of the data point. Curve No. 1 is the theoretical playback time curve of the data point, and Curve No. 1 is the change curve of the playback delay of the data point represented by No. 2 and No. 3 curves.
1、突发延迟测试数据集1. Burst delay test data set
突发延迟测试数据集即带有延迟峰值的测试数据集,在一个点上延迟突然增加,然后快速下降。这种测试数据集使得接收设备即待测的多媒体终端突然收不到RTP包,然后是很多紧挨着的包,可以测试多媒体终端对延迟突变的适应性。从图2中可以看出,由于突发延迟的存在,一段时间内终端播放缓冲区没有输入包,原来缓存的包逐渐消耗掉,如果突发延迟大到超过缓冲区缓存时间,就可以从图5的波形中测量出缓冲区的缓存能力。Burst latency test datasets are test datasets with latency spikes, at one point the latency suddenly increases and then drops rapidly. This test data set makes the receiving device, that is, the multimedia terminal to be tested, unable to receive RTP packets suddenly, and then a lot of packets next to each other, which can test the adaptability of the multimedia terminal to delay mutations. It can be seen from Figure 2 that due to the existence of the burst delay, the terminal playback buffer has no input packets for a period of time, and the original cached packets are gradually consumed. If the burst delay is too large to exceed the buffer cache time, the The cache capacity of the buffer is measured in the waveform of 5.
2、振荡延迟测试数据集2. Oscillation delay test data set
振荡延迟可以暴露一些设计不是很好的QoS算法不适应的方面。从图3中可以看出,由于输入音频数据附加了一个振荡,对于某些播放缓冲区播放算法,会导致性能的恶化,因而可以测试出一些自适应QoS算法的鲁棒性。Oscillation delays can expose aspects that poorly designed QoS algorithms do not adapt to. It can be seen from Fig. 3 that because an oscillation is added to the input audio data, the performance will deteriorate for some playback buffer playback algorithms, so the robustness of some adaptive QoS algorithms can be tested.
3、步进延迟测试数据集3. Step delay test data set
音频数据在播放缓冲区的缓冲,这段时间一般不容易估计。若用一延迟缓慢步进增加的测试数据集,在缓冲能力内延迟就会被吸收,但在接近缓冲区能力的时候,由于丢包补偿的作用,此时的延迟时间就跟缓冲区能力最接近了。从图5中可以看出,随着输入音频延迟逐步增加,最终在音频输入延迟接近缓冲区能力的时候,播放缓冲区的输出波形会变得不连续。这个测试集比突发延迟测试集更有效地测定了播放缓冲的QoS能力。Audio data is buffered in the playback buffer, and this period of time is generally not easy to estimate. If you use a test data set with a delay that increases slowly, the delay will be absorbed within the buffer capacity, but when it is close to the buffer capacity, due to the effect of packet loss compensation, the delay time at this time will be the best with the buffer capacity. close. It can be seen from Figure 5 that as the input audio delay gradually increases, when the audio input delay is close to the buffer capacity, the output waveform of the playback buffer will become discontinuous. This test set measures the QoS capability of playback buffering more effectively than the Burst Delay test set.
4、延迟跳变测试数据集4. Delay jump test data set
另一种测定播放缓冲最大能力的方法是利用延迟跳变测试数据集。假设待测的多媒体终端的播放缓冲区不超过80ms,则利用一200ms的延迟跳变测试数据集来测试。由于一个200ms的延迟,当缓冲的包播放完成后,待测的多媒体终端会启动丢包补偿,音频渐渐变小,然后当新的RTP包到来后,部分会被丢弃,剩余部分在缓冲范围内的会被立即播放,那么从启动丢包补偿到延迟的音频播放,我们可以从播放的波形推断出这一时间。如图4所示,W1为原始波形,W2为丢包补偿波形,W3为输出波形,SW为延迟阶跃,Te为波形从产生到传播至多媒体终端播放出来的延迟时间,即通常所说的嘴到耳时间,则播放缓冲区最大缓冲能力为200-Td毫秒。Another way to determine the maximum capability of playback buffering is to use the latency jump test data set. Assuming that the playback buffer of the multimedia terminal to be tested does not exceed 80ms, a 200ms delay jump test data set is used for testing. Due to a 200ms delay, when the buffered packets are played, the multimedia terminal under test will start packet loss compensation, the audio will gradually become smaller, and then when a new RTP packet arrives, part of it will be discarded, and the rest will be within the buffer range will be played immediately, then from the start of packet loss compensation to the delayed audio playback, we can infer this time from the played waveform. As shown in Figure 4, W1 is the original waveform, W2 is the packet loss compensation waveform, W3 is the output waveform, SW is the delay step, and Te is the delay time from the generation of the waveform to the broadcasting of the multimedia terminal, which is commonly referred to as For the mouth-to-ear time, the maximum buffering capacity of the playback buffer is 200-Td milliseconds.
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| CN1802639A (en) * | 2003-06-13 | 2006-07-12 | 苹果电脑公司 | Synchronized transmission of audio and video data from a computer to a client via an interface |
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