Also fix the bytestream reader size parameter to take the
offset into account.

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

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

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

Set the transactionId to 0 to signal the server a return _result/_error
is not expected.

Solves the problems experienced with strict rtmp servers.

This option specifies the name of live stream to subscribe.
Defaults to rtmp_playpath.

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

When streaming live streams using the Akamai, Edgecast or Limelight CDN,
players cannot simply connect to the live stream. Instead, they have to
subscribe to it, by sending an FC Subscribe call to the server.

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

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

They are managed in get_packet()

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

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

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

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

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

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

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

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

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

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

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

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

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

This option specifies the URL of the web page in which the media
was embedded.

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

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

This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.

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

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

This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// 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: Martin Storsjö <martin@martin.st>

This function is used for calculating digest position for RTMP handshake
packets.

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

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

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

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

The prefix makes it easier to distinguish the proper end-user
protocols from the internal ones.

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

Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>

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: Martin Storsjö <martin@martin.st>

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: Martin Storsjö <martin@martin.st>

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: Martin Storsjö <martin@martin.st>

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

This factorizes existing code into a new function gen_buffer_time(),
which generates the client buffer time message and sends it to the
server.

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

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

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

This fixes builds on platforms without strtok_r (windows).

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

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

This fixes bugzilla bug #304.

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

Allow using connection parameters in order to append arbitrary
AMF data like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0" to the
Connect message. You can pass these parameters through the -rtmp_conn
option.

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

The application component can have a subcomponent to specify the
application instance even if it doesn't have a ":" in the playpath.

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

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

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

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

Check malloc calls used by ff_rtmp_packet_create, unify error
handling and pass on error codes.

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