This adds a GStreamer-based RTP implementation to replace our own. The
original implementation is retained for cases where it is not possible
to include GStreamer as a dependency.
The idea with this is to be able to start supporting more advanced RTP
features such as RTCP, non-PCM audio, and potentially synchronised
playback.
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
It is possible that we might want to have a separate userdata to be used
for these callbacks, so let's split them out.
This is particularly needed when using an pa_rtpoll_item around pa_fdsem
since that uses its own before/after callback but will essentially have
whatever is using the fdsem set up the work callback appropriately (and
thus at least the work callback's userdata needs to be separated from
the before/after callback -- we might as well then just separate all
three).
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
This moves RTP implementation-specific information out of
module-rtp-send/recv. This is basically done by making the
pa_rtp_context structure opaque from the perspective of these modules.
We can then potentially replace the underlying RTP implementation with
something else transparently.
One RTP detail that does "leak" is the RTP timestamp. We provide this to
module-rtp-recv so that it can perform rate adjustments to match the
sender rate.
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
module-rtp-send itself doesn't really need to handle this, the
implementation can keep track (and make sure sending happens in MTU
sized chunks).
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
There doesn't seem much value in supporting streaming U8/mulaw/alaw on
the network, and it's unlikely these get any testing. Makes more sense
to drop these formats and just convert to L16 if we're dealing with
source media in that format.
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
This was being done automatically by autotools, now we need to manually
specify this for each executable/library with a dependency in a
non-standard directory.
Brings things in line with the autotools build, and adds ALSA mixer
paths and profile-sets into the meson build system as well.
The module installation path is also now customisable.
Add configuration option 'stream_name' for stream/session name so user
will see it on receiver side as RTP Strean ($stream_name)
ex: load-module module-rtp-send source=rtp.monitor stream_name=MyServerMedia
This removes the symdef header generation m4 magic in favour of a
simpler macro method, allowing us to skip one unnecessary build step
while moving to meson, and removing an 11 year old todo!
When a stream is created, and the stream creator specifies which device
should be used, that can affect automatic routing policies.
Specifically, module-device-manager shouldn't apply its priority list
routing when a stream has been routed by the application that created
the stream.
A stream that was initially routed by the application may be moved for
some valid reason (e.g. user requesting a move, or the original device
disappearing). When the stream is moved away from its initial device,
the "device requested by application" flag isn't relevant any more, so
it's set to false and never reset to true again.
The change in module-device-manager's routing logic will be done in the
following patch.
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.
