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Merge remote-tracking branch 'qatar/master'
* 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:
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@ -258,6 +258,32 @@ git commit
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@end example
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@chapter Git configuration
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In order to simplify a few workflows, it is advisable to configure both
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your personal Git installation and your local FFmpeg repository.
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@section Personal Git installation
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Add the following to your @file{~/.gitconfig} to help @command{git send-email}
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and @command{git format-patch} detect renames:
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@example
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[diff]
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renames = copy
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@end example
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@section Repository configuration
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In order to have @command{git send-email} automatically send patches
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to the ffmpeg-devel mailing list, add the following stanza
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to @file{/path/to/ffmpeg/repository/.git/config}:
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@example
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[sendemail]
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to = ffmpeg-devel@@ffmpeg.org
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@end example
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@chapter FFmpeg specific
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@section Reverting broken commits
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@ -2892,7 +2892,7 @@ AVCodec ff_aac_decoder = {
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.init = aac_decode_init,
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.close = aac_decode_close,
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.decode = aac_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
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.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
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.sample_fmts = (const enum AVSampleFormat[]) {
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
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},
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@ -398,6 +398,7 @@ OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o
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OBJS-$(CONFIG_TLS_PROTOCOL) += tls.o
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OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
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SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
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SKIPHEADERS-$(CONFIG_NETWORK) += network.h rtsp.h
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TESTPROGS = seek
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@ -1,5 +1,7 @@
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/*
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* RTMPE network protocol
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* Copyright (c) 2008-2009 Andrej Stepanchuk
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* Copyright (c) 2009-2010 Howard Chu
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* Copyright (c) 2012 Samuel Pitoiset
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*
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* This file is part of FFmpeg.
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@ -1,5 +1,7 @@
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/*
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* RTMP Diffie-Hellmann utilities
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* Copyright (c) 2009 Andrej Stepanchuk
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* Copyright (c) 2009-2010 Howard Chu
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* Copyright (c) 2012 Samuel Pitoiset
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*
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* This file is part of FFmpeg.
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@ -719,7 +719,6 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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int i;
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int server_pos, client_pos;
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uint8_t digest[32], signature[32];
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#define encrypted (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted)
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int ret, type = 0;
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av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
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@ -729,7 +728,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
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tosend[i] = av_lfg_get(&rnd) >> 24;
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if (encrypted) {
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if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
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/* When the client wants to use RTMPE, we have to change the command
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* byte to 0x06 which means to use encrypted data and we have to set
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* the flash version to at least 9.0.115.0. */
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@ -745,7 +744,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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return ret;
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}
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client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, encrypted);
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client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
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if (client_pos < 0)
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return client_pos;
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@ -799,7 +798,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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if (ret < 0)
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return ret;
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if (encrypted) {
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if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
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/* Compute the shared secret key sent by the server and initialize
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* the RC4 encryption. */
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if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
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@ -829,7 +828,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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if (ret < 0)
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return ret;
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if (encrypted) {
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if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
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/* Encrypt the signature to be send to the server. */
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ff_rtmpe_encrypt_sig(rt->stream, tosend +
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RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
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@ -841,13 +840,13 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
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return ret;
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if (encrypted) {
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if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
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/* Set RC4 keys for encryption and update the keystreams. */
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if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
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return ret;
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}
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} else {
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if (encrypted) {
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if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
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/* Compute the shared secret key sent by the server and initialize
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* the RC4 encryption. */
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if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
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@ -865,7 +864,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
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return ret;
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if (encrypted) {
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if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
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/* Set RC4 keys for encryption and update the keystreams. */
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if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
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return ret;
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@ -874,7 +873,6 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
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return 0;
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}
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#undef encrypted
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/**
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* Parse received packet and possibly perform some action depending on
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!fmt->oformat->priv_class ||
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!av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
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continue;
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if (codec->codec_id == CODEC_ID_PCM_S16BE)
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if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
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continue;
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/* G722 has 8000 as nominal rate even if the sample rate is 16000,
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* see section 4.5.2 in RFC 3551. */
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if (codec->codec_id == CODEC_ID_ADPCM_G722 &&
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codec->sample_rate == 16000 && codec->channels == 1)
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return AVRtpPayloadTypes[i].pt;
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if (codec->codec_type == AVMEDIA_TYPE_AUDIO &&
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((AVRtpPayloadTypes[i].clock_rate > 0 &&
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codec->sample_rate != AVRtpPayloadTypes[i].clock_rate) ||
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(AVRtpPayloadTypes[i].audio_channels > 0 &&
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codec->channels != AVRtpPayloadTypes[i].audio_channels)))
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continue;
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return AVRtpPayloadTypes[i].pt;
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}
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@ -1,5 +1,5 @@
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484aeef3be3eb4deef05c83bdc2dd484 *./tests/data/lavf/lavf.mov
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367346 ./tests/data/lavf/lavf.mov
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2c58e6422013d9cd6169f272ffba7a33 *./tests/data/lavf/lavf.mov
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367373 ./tests/data/lavf/lavf.mov
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./tests/data/lavf/lavf.mov CRC=0x2f6a9b26
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21b992f6a677f971dfd685cc055a2b0a *./tests/data/lavf/lavf.mov
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358463 ./tests/data/lavf/lavf.mov
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