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Code Issues Releases Activity

Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
  rtmp: Move the CONFIG_ condition into the if conditions
  aac: Mention abbreviation as well in long_name
  build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
  doc: Add Git configuration section
  configure: Add a dependency on https for rtmpts
  rtp: Only choose static payload types if the sample rate and channels are right

Conflicts:
	doc/git-howto.texi
	libavformat/rtmpproto.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-07-24 20:43:07 +02:00
commit 93342de1d8
8 changed files with 52 additions and 15 deletions
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26
doc/git-howto.texi
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@ -258,6 +258,32 @@ git commit
@end example
@chapter Git configuration
In order to simplify a few workflows, it is advisable to configure both
your personal Git installation and your local FFmpeg repository.
@section Personal Git installation
Add the following to your @file{~/.gitconfig} to help @command{git send-email}
and @command{git format-patch} detect renames:
@example
[diff]
renames = copy
@end example
@section Repository configuration
In order to have @command{git send-email} automatically send patches
to the ffmpeg-devel mailing list, add the following stanza
to @file{/path/to/ffmpeg/repository/.git/config}:
@example
[sendemail]
to = ffmpeg-devel@@ffmpeg.org
@end example
@chapter FFmpeg specific
@section Reverting broken commits

2
libavcodec/aacdec.c
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@ -2892,7 +2892,7 @@ AVCodec ff_aac_decoder = {
.init = aac_decode_init,
.close = aac_decode_close,
.decode = aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},

1
libavformat/Makefile
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@ -398,6 +398,7 @@ OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o
OBJS-$(CONFIG_TLS_PROTOCOL) += tls.o
OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
SKIPHEADERS-$(CONFIG_NETWORK) += network.h rtsp.h
TESTPROGS = seek

2
libavformat/rtmpcrypt.c
View File

@ -1,5 +1,7 @@
/*
* RTMPE network protocol
* Copyright (c) 2008-2009 Andrej Stepanchuk
* Copyright (c) 2009-2010 Howard Chu
* Copyright (c) 2012 Samuel Pitoiset
*
* This file is part of FFmpeg.

2
libavformat/rtmpdh.c
View File

@ -1,5 +1,7 @@
/*
* RTMP Diffie-Hellmann utilities
* Copyright (c) 2009 Andrej Stepanchuk
* Copyright (c) 2009-2010 Howard Chu
* Copyright (c) 2012 Samuel Pitoiset
*
* This file is part of FFmpeg.

16
libavformat/rtmpproto.c
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@ -719,7 +719,6 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
int i;
int server_pos, client_pos;
uint8_t digest[32], signature[32];
#define encrypted (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted)
int ret, type = 0;
av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
@ -729,7 +728,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
tosend[i] = av_lfg_get(&rnd) >> 24;
if (encrypted) {
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
/* When the client wants to use RTMPE, we have to change the command
* byte to 0x06 which means to use encrypted data and we have to set
* the flash version to at least 9.0.115.0. */
@ -745,7 +744,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
return ret;
}
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, encrypted);
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
if (client_pos < 0)
return client_pos;
@ -799,7 +798,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
if (ret < 0)
return ret;
if (encrypted) {
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
/* Compute the shared secret key sent by the server and initialize
* the RC4 encryption. */
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
@ -829,7 +828,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
if (ret < 0)
return ret;
if (encrypted) {
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
/* Encrypt the signature to be send to the server. */
ff_rtmpe_encrypt_sig(rt->stream, tosend +
RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
@ -841,13 +840,13 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
return ret;
if (encrypted) {
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
/* Set RC4 keys for encryption and update the keystreams. */
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
return ret;
}
} else {
if (encrypted) {
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
/* Compute the shared secret key sent by the server and initialize
* the RC4 encryption. */
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
@ -865,7 +864,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
return ret;
if (encrypted) {
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
/* Set RC4 keys for encryption and update the keystreams. */
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
return ret;
@ -874,7 +873,6 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
return 0;
}
#undef encrypted
/**
* Parse received packet and possibly perform some action depending on

12
libavformat/rtp.c
View File

@ -110,8 +110,16 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec)
!fmt->oformat->priv_class ||
!av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
continue;
if (codec->codec_id == CODEC_ID_PCM_S16BE)
if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
/* G722 has 8000 as nominal rate even if the sample rate is 16000,
* see section 4.5.2 in RFC 3551. */
if (codec->codec_id == CODEC_ID_ADPCM_G722 &&
codec->sample_rate == 16000 && codec->channels == 1)
return AVRtpPayloadTypes[i].pt;
if (codec->codec_type == AVMEDIA_TYPE_AUDIO &&
((AVRtpPayloadTypes[i].clock_rate > 0 &&
codec->sample_rate != AVRtpPayloadTypes[i].clock_rate) ||
(AVRtpPayloadTypes[i].audio_channels > 0 &&
codec->channels != AVRtpPayloadTypes[i].audio_channels)))
continue;
return AVRtpPayloadTypes[i].pt;
}

4
tests/ref/lavf/mov
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@ -1,5 +1,5 @@
484aeef3be3eb4deef05c83bdc2dd484 *./tests/data/lavf/lavf.mov
367346 ./tests/data/lavf/lavf.mov
2c58e6422013d9cd6169f272ffba7a33 *./tests/data/lavf/lavf.mov
367373 ./tests/data/lavf/lavf.mov
./tests/data/lavf/lavf.mov CRC=0x2f6a9b26
21b992f6a677f971dfd685cc055a2b0a *./tests/data/lavf/lavf.mov
358463 ./tests/data/lavf/lavf.mov