Hi Guedes,
On Aug 29 2018 10:00, Guedes, Andre wrote:
On Sat, 2018-08-25 at 17:13 +0900, Takashi Sakamoto wrote:
On Aug 24 2018 03:32, Guedes, Andre wrote:
On Wed, 2018-08-22 at 21:25 -0500, Pierre-Louis Bossart wrote:
On 08/22/2018 07:46 PM, Guedes, Andre wrote:
On Tue, 2018-08-21 at 17:51 -0500, Pierre-Louis Bossart wrote:
>>> +static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf) >>> +{ >>> + int res; >>> + struct timespec now; >>> + struct itimerspec itspec; >>> + snd_pcm_ioplug_t *io = &aaf->io; >>> + >>> + res = clock_gettime(CLOCK_REF, &now); >>> + if (res < 0) { >>> + SNDERR("Failed to get time from >>> clock"); >>> + return -errno; >>> + } >>> + >>> + aaf->mclk_period = (NSEC_PER_SEC * aaf- >>>> frames_per_pkt) / >>> >>> io->rate; >> >> is this always an integer? If not, don't you have a >> systematic >> arithmetic error? > > NSEC_PER_SEC is 64-bit so I don't see an arithmetic error > during > calculation (e.g. integer overflow). Not sure this was your > concern, > though. Let me know otherwise.
No, I was talking about the fractional part, e.g with 256 frames with 44.1kHz you have a period of 5804988.662131519274376 - so your math adds a truncation. same with 48khz, the fractional part is .333
I burned a number of my remaining neurons chasing a <100 ppb error which led to underruns after 10 hours, so careful now with truncation...
Thanks for clarifying.
Yes, we can end up having a fractional period which is truncated. Note that both 'frames' and 'rate' are configured by the user. The user should set 'frames' as multiple of 'rate' whenever possible to avoid inaccuracy.
It's unlikely to happen. it's classic in audio that people want powers of two for fast filtering, and don't really care that the periods are fractional. If you cannot guarantee long-term operation without timing issues, you should add constraints to the frames and rates so that there is no surprise.
Fair enough. So for now I'll add a constraint on frames and rates to unsure no surprises. Later we can revisit this and implement the compesation mechanism you described below.
In my understanding, transmission timing of 'AVTP Audio format' in IEEE 1722:2016 is similar to 'blocking transmission' of IEC 61883-6. Packets have fixed size of data in its payload, thus include the same number of PCM frames. Talkers are expected to fill data till the size, then transmit the packet. Receivers are expected to perform buffering till presentation timestamp is elapsed, with one (or more) AVTPDUs.
I'm not familiar with the 'blocking transmission' of IEC 61883-6 but from the description above, yes, it looks similar indeed.
Further investigation, I realized that transmission timing of AAF is not similar to IEC 61883-1/6 at all... I'm sorry to address to it into this topic.
In clause 7.7 'AAF and SRP', I can see below sentence: '... A 44.1-kHz stream with 6 samples per AAF AVTPDU and an FQTSS observation interval of 125 us also has an SRP reservation of 1 frame per observation interval even though there will periodically be an observation interval where no AAF AVTPDU will be transmitted since it has a transmission interval of 136.054 us as can be seen in the example given in Figure 36.' This means that packet transmission is not always periodically. There's a blank cycle per several cycles; like IEC 61883-1/6.
My understanding of the periodicity of packet transmission is different. I believe it is always periodic. Let me elaborate on this.
The first paragraph from Section 7.7 states that "AAF transmission interval is defined by the clock rate of the media rather than the FQTSS observation interval." Since the clock is periodic, the AAF transmission interval is periodic too. For instance, if we take the 44.1 kHz example from Figure 36, we can see the AAF transmission interval is always ~136us. So, from AAF perspective (i.e. the plugin perspective), the packet transmission is always periodic.
The AAF transmission interval isn't necessarily equal to the FQTSS observation interval. Again, if we take a look at the 44.1 kHz example, we can see the AAF transmission interval is ~136us while the FQTSS observation interval is 125us. This, however, isn't an issue since the plugin is not expected to operate in terms of FQTSS observation interval, but in terms of AAF transmission interval as stated in the first paragraph from Section 7.7.
Indeed, thanks for your correction against my misunderstanding.
In my opinion, it's better calculate proper interval of timerfd to create the black interval, without truncate the fraction. Then, give proper constrains to SND_PCM_IOPLUG_HW_PERIOD_BYTES to prevent applications from underrun.
If the above understanding is correct, I'm not sure this approach would work. Let me know otherwise.
I have another concern of buffering in a perspectives of delay of task scheduling.
The interval of task scheduling for this plugin is decided mainly by the value of 'frames_per_pkt', given by users. In your documentation, the value is 6[1]. Of cource this is an example but in this case the interval is calculated as 125us at 48.0kHz. In my opinion, task scheduling in Linux kernel brings deadline misses for the interval, in most cases such as major Linux distribution on usual personal computers. When considering about the fact that recent motherboards implements Intel I210/220 series, it's better to care for the low-level realtime systems, in my opinion.
Furthermore, in your proposal, the number of PCM frames in one AVTPDU is decided according to plugin parameter. However, if compliant to specification, it's better to decide the number according to 'FQTSS observation interval'. I can see recommendations in two cases; FQTSS = 125 us and 256 us, in Table 17 and 18 of IEEE 1212:2016.
It was designed that way on purpose. Let me share the rationale.
The values in Table 17 and 18 are just recommendations. From the plugin perspective, we should enable users to configure the number of frames according to their needs. This way, users are free to configure the values recommended by the spec or other values optimized to their AVTP application.
Besides that, as stated in the NOTE right below Table 18, 125us and 250us are not the only possible FQTSS observation intervals.
I addressed to the reccomendation itself. If specification describes recommendations, there will be a reason to consider about it, just not refer to values on the table.
I can see a sentence in section 7.7; 'in order to maintain interoperability between devices, the transmission intervals listed in Table 17 and Table 18 should be used.' I understand that 'if a talker end station transfers an AVTP packet with largely different number of sample frames from expectations on listener end station, there's a case that the listener cannot handles the sample frames due to processor loading or buffer overflow on listener side. To avoid this case, FQTSS observation interval is loosely used to decide intervals of AVTPDU transmission'. But it's within my imagination.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139495.html
Thanks
Takashi Sakamoto