Hi,
Patrick Mackin wrote:
>> Regarding radio recordings, the biggest problem is, that there is no
>> video stream which would generate an index entry every 40 ms (for the
>> default value of 25 frames per second). Since the introduction of my
>> cAudioRepacker, things got even worse but more predictable: now every
>> audio frame generates an index entry.
>
> Unfortunately I don't have any ideas on how to fix the problem with
> every audio frame generating an index entry, and changing the method
> that vdr calculates the time of a recording would be a major
> undertaking. I don't use HDTV at all, but from what I understand,
> something is "broken" there as well, that may require a substantial
> change in how vdr calculates the time of a recording. Progressive video
> (ie 720p) uses twice the framerate of interlaced (standard NTSC or PAL)
> so this may be the source of the problem. Also, as I mentioned, some
> north american providers [SDTV] use a variable frame rate *during*
> playback of a small percentage of programming, which causes the reported
> time to be less than the actual recording time (by approximately 20% in
> my tests). I would assume this is done for bandwidth optimization
> reasons. I'm also guessing it is not allowed by dvb-s standards, and
> therefore may not be an issue that affects enough people for it to be
> worth fixing.
I'd like to have a look at such recordings. Can you provide me some URLs?
Regarding HDTV: I'm currently experimenting with H.264 coded 1080i
streams and what I've found so far is that the encoder may choose on the
fly whether to use a single frame or two fields for a picture. But it
looks like I've found a way (at least for this single stream) which will
suit to VDR's way of determining playback times.
>> But as the duration of an audio frame depends on bitrate, sample rate,
>> and further audio stream parameters, it typically isn't 40 ms, but
>> should be constant until the stream parameters change. That's why you
>> get a consistent error for playback times of radio recordings.
>
> What is interesting is that on 64kb, 96 kb and 128kb audio channels, the
> recording times were all basically identical. But perhaps you are
> saying that only if a stream switches from say 64 to 96 kb *during* a
> recording would this affect the recording times.
I've found this code in my vdr-xine plugin to determine the duration of
an audio frame in PTS (90000 PTS == 1 second):
static int samplingFrequencies[ 2 ][ 4 ] = // all values are
specified in Hz
{
{ 44100, 48000, 32000, -1 }, // MPEG 1
{ 22050, 24000, 16000, -1 } // MPEG 2
};
static int slots_per_frame[ 2 ][ 3 ] =
{
{ 12, 144, 144 }, // MPEG 1, Layer I, II, III
{ 12, 144, 72 } // MPEG 2, Layer I, II, III
};
int mpegIndex = 1 - id;
int layerIndex = 3 - layer;
// Layer I (i. e., layerIndex == 0) has a larger slot size
int slotSize = (layerIndex == 0) ? 4 : 1; // bytes
int sf = samplingFrequencies[ mpegIndex ][ sampling_frequency ];
return 90000.0 * 8 * slotSize * slots_per_frame[ mpegIndex ][
layerIndex ] / sf;
In the case of MPEG 1 Layer II, there is only "sf" which varies with the
sampling frequency (sampling_frequency is a 2 bit field in the audio
frame's header). But as there is no dependency on bitrate_index (a 4 bit
field in the audio frame's header), I was simply wrong :-(
>> I've already thought about patching VDR that it determines FRAMESPERSEC
>> dynamically by looking at the duration of the first frame of a
>> recording, but I've dropped that idea as there are channels which
>> provide more than one radio stream. So, to fix this issue I'd have to
>> fix cRemux to not generate an index entry for every audio frame. I have
>> an idea now how to achieve this -- don't know why it didn't come into my
>> mind earlier already ;-)
>
> My only idea would be to perhaps use vdr's internal clock for timing, as
> that would produce correct results regardless of the content recorded,
> but that would involve a substantial rewrite and may have some
> unforeseen disadvantages that I'm unaware of. I don't quite understand
> the solution you mention here--you say you could fix cRemux to not
> generate an index entry for every audio frame (as currently audio frames
> are generated faster than every 40ms) but then what would the timer be
> based on? I'm not familiar with how regular audio players interpret the
> time of a recording, but it would be interesting to see how audacity or
> an open source audio program would interpret the time of the very same
> recording vdr incorrectly interprets.
Well my idea is to simply accumulate the above calculated frame
durations and whenever the result exceeds a multiple of 40 ms (for 25
frames per second), an index entry is generated. To be more precise the
PTS of the audio PES packets needs to be considered too but that's a
little bit too complicated to explain.
Bye.
--
Dipl.-Inform. (FH) Reinhard Nissl
mailto:rnissl@xxxxxx
