The reported latency of source or sink is based on measured initial conditions.
If the conditions contain an error, the estimated latency values may become negative.
This does not indicate that the latency is indeed negative but can be considered
merely an offset error. The current get_latency_in_thread() calls and the
implementations of the PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY messages truncate negative
latencies because they do not make sense from a physical point of view. In fact,
the values are truncated twice, once in the message handler and a second time in
the pa_{source,sink}_get_latency_within_thread() call itself.
This leads to two problems for the latency controller within module-loopback:
- Truncating leads to discontinuities in the latency reports which then trigger
unwanted end to end latency corrections.
- If a large negative port latency offsets is set, the reported latency is always 0,
making it impossible to control the end to end latency at all.
This patch is a pre-condition for solving these problems.
It adds a new flag to pa_{sink,source}_get_latency_within_thread() to allow
negative return values. Truncating is also removed in all implementations of the
PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY message handlers. The allow_negative flag
is set to false for all calls of pa_{sink,source}_get_latency_within_thread()
except when used within PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY. This means that the
original behavior is not altered in most cases. Only if a positive latency offset
is set and the message returns a negative value, the reported latency is smaller
because the values are not truncated twice.
Additionally let PA_SOURCE_MESSAGE_GET_LATENCY return -pa_sink_get_latency_within_thread()
for monitor sources because the source gets the data before it is played.
The RTSP client is not waiting anymore a new header after the
previous one (which can never occurs if RAOP is disconnected)
but after sending a command.
This patch fixes Issue #36.
https://github.com/hfujita/pulseaudio-raop2/issues/36
TCP and UDP implementation are following two diffrent code path while code
logic is quite the same. This patch merges both code path into a unique one
and, thus, leads to a big refactoring. Major changes include:
- moving sink implementation to a separate file (raop-sink.c)
- move raop-sink.c protocol specific code to raop-client.c
- modernise RTSP session handling in TCP mode
- reduce code duplications between TCP and UDP modes
- introduce authentication support
- TCP mode does not constantly send silent audio anymore
About authentication: OPTIONS is now issued when the sink is preliminary
loaded. Client authentication appends at that time and credential is kept
for the whole sink lifetime. Later RTSP connection will thus look like this:
ANNOUNCE > 200 OK > SETUP > 200 OK > RECORD > 200 OK (no more OPTIONS). This
behaviour is similar to iTunes one.
Also this patch includes file name changes to match Pulseaudio naming
rules, as most of pulseaudio source code files seem to be using '-'
instead of '_' as a word separator.
RAOP authentication is using standard HTTP challenge-response authentication
scheme. This patch adds two helper functions that generate the proper hash
(for both techniques) given a username, a password and session related tokens.
In the current RTSP implementation, there is a vulnerable window
between the RTSP object creation and the URL initialization.
If any RTSP command is issued during this period, it will lead to
crash by assertion violation.
This patch introduces pa_rtsp_exec_ready(), which returns if it is
safe to issue RTSP commands.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
Add a function performing a call to the OPTIONS request; also,
in some special cases, tuning transport parameters is required (default:
"RTP/AVP/TCP;unicast;interleaved=0-1;mode=record") ! The RAOP client for
example needs to overwrite them.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
pa_ioline_close does not free the ioline structure itself, so we
have to unref the structure if we want to free it.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
pa_memblockq_push() expects all memchunks to be aligned to PCM frame
boundaries, and that means both the index and length fields of
pa_memchunk. pa_rtp_recv(), however, used a memblock for storing both
the RTP packet metadata and the actual audio data. The metadata was
"removed" from the audio data by setting the memchunk index
appropriately, so the metadata stayed in the memblock, but it was not
played back. The metadata length is not necessarily divisible by the PCM
frame size, which caused pa_memblock_push() to crash in an assertion
with some sample specs, because the memchunk index was not properly
aligned. In my tests the metadata length was 12, so it was compatible
with many configurations, but eight-channel audio didn't work.
This patch adds a separate buffer for receiving the RTP packets. As a
result, an extra memcpy is needed for moving the audio data from the
receive buffer to the memblock buffer.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=96612
The code was mixing sink and sink input domain rate updates, and that
only works if the rate of the RTP stream is the same as the rate of the
sink. This changes all the calcuations to be on the sink-input rate,
since that's the rate we are trying to guess (and resample for).
In rtp.c:
if (sscanf(t+9, "%i %64c", &_payload, c) == 2)
the string c seems to be non-null terminated. It is later used as
following:
c[strcspn(c, "\n")] = 0;
The same piece of code is responsible for the inability of pulseaudio
on OpenWRT to handle RTP stream at the rate 48000 from another
machine:
[pulseaudio] sdp.c: Failed to parse SDP data: missing data.
It turns out that uClibc does not agree with glibc about "%64c", see
http://git.uclibc.org/uClibc/tree/docs/Glibc_vs_uClibc_Differences.txt
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=92568
FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
On FIONREAD returning 0 bytes, we cannot return success, as the caller
(rtpoll_work_cb in module-rtp-recv.c) would then try to
pa_memblock_unref(chunk.memblock) and, because memblock is NULL, trigger
an assertion.
Also we have to read out the possible empty packet from the socket, so
that the kernel doesn't tell us again and again about it.
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
Some people want module-rtp-send to send silence when the sink that is
monitored goes idle, and some people want module-rtp-send to pause the
RTP stream to avoid unnecessary bandwidth consumption.
Since the hashmap stores a pointer to the key provided at pa_hashmap_put()
time, it make sense to allow the hashmap to be given ownership of the key and
have it free it at pa_hashmap_remove/free time.
To do this cleanly, we now provide the key and value free functions at hashmap
creation time with a pa_hashmap_new_full. With this, we do away with the free
function that was provided at remove/free time for freeing the value.
This patch removes all occurrences of double and triple
newlines.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name 'adrian-aec.*' -a -not \
-name reserve.c -a -not -name 'rtkit.*' \
-exec sed -i -e '/^$/{N;s/^\n$//}' {} \;
Two passes were needed to remove triple newlines.
The excluded files are mirrored files from external sources.
On a multi-homed system, the user may wish RTP to be used only on
specific interfaces. The default binding of 0.0.0.0 for the source
address causes SAP multicast on all interfaces, which is not ideal.
Introduce a new module argument, that allows selection of the source IP,
and thus interface.
(changes in v2: s/srcip/source_ip)
Signed-off-by: Robin H. Johnson <robbat2@gentoo.org>
The module argument 'source' already has special meaning as the
pa_source, however, the argument 'destination' expects an IP address.
Prior to introducing a source IP modarg for the source IP address,
rename the 'destination' argument to 'destination_ip'. Include
compatibility support for old RTP users so they don't need to change
their module usage immediately.
(changes in v2: minor formatting fixes, s/dstip/destination_ip)
Signed-off-by: Robin H. Johnson <robbat2@gentoo.org>
Before introducing new functionality, clarify the variable names
dest -> dst_addr
sa[46] -> dst_sa[46]
sap_sa[46] -> dst_sap_sa[46]
Signed-off-by: Robin H. Johnson <robbat2@gentoo.org>
The previous patch removed module-gconf's dependency on the userdata
pointer of the free callback, and that was the only place where the
userdata pointer of pa_free2_cb_t was used, so now there's no need for
pa_free2_cb_t in pa_hashmap_free(). Using pa_free_cb_t instead allows
removing a significant amount of repetitive code.
modules/rtp/module-rtp-recv.c:462:8: warning: 'r' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
These are not used for anything at this point, but this
makes it easy to add ad-hoc debug prints that show the
memblockq name and to convert between bytes and usecs.
The Apple TV for example uses a non-default port, but we previously ignored
this. We now correctly parse the server string but in so doing, we end up
parsing the address twice. As we need a pure IP/hostname of the device itself
to use in our requests, this is somewhat unavoidable.
Sadly there are still other problems with Apple TVs, but this is still
one step closer.
Fixes part of #950
This is the beginning of work to support compressed formats natively in
PulseAudio. This adds a pa_stream_new_extended() that takes a format
structure, sends it to the server (=> protocol extension) and has the
server negotiate with the appropropriate sink to figure out what format
it should use.
This is work in progress, and works only with PCM streams. Actual
compressed format support in some sink needs to be implemented, and
extensive testing is required.
More details on how this is supposed to work is available at:
http://pulseaudio.org/wiki/PassthroughSupport
If the virtual sink is moved to a new master right after it has been created,
then the virtual sink input's memblockq can be rewound to a negative read
index. The data written prior to the move starts from index zero, so after the
rewind there's a bit of silence. If the memblockq doesn't have a silence
memchunk set, then pa_memblockq_peek() will return zero in such case, and the
returned memchunk's memblock pointer will be NULL.
That scenario wasn't taken into account in the implementation of
sink_input_pop_cb. Setting a silence memchunk for the memblockq solves this
problem, because pa_memblock_peek() will now return a valid memblock if the
read index happens to point to a hole in the memblockq.
I believe this isn't the best possible solution, though. It doesn't really make
sense to rewind the sink input's memblockq beyond index 0 in the first place,
because now when the stream starts to play to the new master sink, there's some
unnecessary silence before the actual data starts. This is a small problem,
though, and I don't grok the rewinding system well enough to know how to fix
this issue properly.
I went through all files that call pa_memblockq_peek() to see if there are more
similar bugs. play-memblockq.c was the only one that looked to me like it might
be broken in the same way. I didn't try reproducing the bug with
play-memblockq.c, though, so I just added a FIXME comment there.
Instead <pulsecore/poll.h> should be included. That file includes poll.h on
platform where it is appropriate. Also remove some unnecessary <ioctl.h>
includes.
