Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMartin Storsjö <martin@martin.st>

The packet size, signalled via block_align, has to be passed via
the container.
Signed-off-by: NMartin Storsjö <martin@martin.st>

The library is 3-clause BSD licensed.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Add a new option 'rtmp_flush_interval' that allows specifying the
number of packets to write before sending it off as a HTTP request.

This is mostly relevant for RTMPT - for plain RTMP, it only controls
how often we check the socket for incoming packets, which shouldn't
affect the performance in any noticeable way.
Signed-off-by: NMartin Storsjö <martin@martin.st>

This seems to be the correct mode to send, according to the
original RTSP RFC, and matches the method RECORD which is
sent later when starting to send data.

Darwin Streaming Server works fine with either of them.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMans Rullgard <mans@mansr.com>

Signed-off-by: NJustin Ruggles <justin.ruggles@gmail.com>

2x speedup of surround decoding on Cortex-A9.
Signed-off-by: NMans Rullgard <mans@mansr.com>

Signed-off-by: NMans Rullgard <mans@mansr.com>

This requires all NAL units to fit within single RTP packets. It
doesn't change the actual packetization for packets that fit, but
errors out and gives a helpful hint if the NAL units would have to
be split, and signals the right packetization mode in the SDP.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Only these three libraries were out of order, the rest was already
neatly sorted.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMans Rullgard <mans@mansr.com>

This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMartin Storsjö <martin@martin.st>

This can happen if doing a new request using the same socket,
but the new request failed, which clears the urlcontext.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMartin Storsjö <martin@martin.st>

This was missed in the previous cleanup patch.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Also return a proper error code in these cases.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Add spaces around operators, fix brace placement and whitespace to
match K&R style, vertically align code, remove redundant != 0 and
convert x == 0 into !x, drop useless braces.
Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMartin Storsjö <martin@martin.st>

Signed-off-by: NMartin Storsjö <martin@martin.st>

This can easily happen when the caller is using a custom AVIOContext.

Behave as if the filename was an empty string in this case.

CC: libav-stable@libav.org

Signed-off-by: NJanne Grunau <janne-libav@jannau.net>

We generally do not declare variables within for statements and
there are compilers that choke on such constructs.

This makes sure all incoming packets are read and handled (and reacted
to) while sending an FLV stream over RTMP to a server. If there were
enough incoming data to fill the TCP buffers, this could potentially
make things block at unexpected places. For the upcoming RTMPT support,
we need to consume all incoming data before we can send the next
request.
Signed-off-by: NMartin Storsjö <martin@martin.st>

While there is no reason for starting a frame with anything else
than a Mode A packet, some senders seem to consistently use Mode B
packets for everything. This fixes depacketization of such streams.
Signed-off-by: NMartin Storsjö <martin@martin.st>