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Merge branch 'develop' into merge/develop
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commit
263ead07f0
4 changed files with 10 additions and 9 deletions
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@ -186,6 +186,7 @@ Other documents:
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> Remark: About the milestone and product plan, please read ([CN][v1_CN_Product], [EN][v1_EN_Product]) wiki.
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<a name="history"></a>
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<a name="changes"></a>
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<a name="change-logs"></a>
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## V5 changes
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2
trunk/3rdparty/st-srs/io.c
vendored
2
trunk/3rdparty/st-srs/io.c
vendored
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@ -112,7 +112,7 @@ int _st_io_init(void)
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/**
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* by SRS, for osx.
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* when rlimit max is negative, for example, osx, use cur directly.
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* @see https://github.com/winlinvip/simple-rtmp-server/issues/336
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* @see https://github.com/ossrs/srs/issues/336
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*/
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if ((int)rlim.rlim_max < 0) {
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_st_osfd_limit = (int)(fdlim > 0? fdlim : rlim.rlim_cur);
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@ -1369,7 +1369,7 @@ class SrsTsHanlder implements ISrsTsHandler
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private function write_h264_ipb_frame(ibps:ByteArray, frame_type:uint, dts:uint, pts:uint, piece:ByteArray):void
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{
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// when sps or pps not sent, ignore the packet.
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/203
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// @see https://github.com/ossrs/srs/issues/203
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if (video_sh_tag.length == 0) {
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return;
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}
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@ -2197,7 +2197,7 @@ class SrsRawAacStream
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// decode the ADTS.
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// @see aac-iso-13818-7.pdf, page 26
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// 6.2 Audio Data Transport Stream, ADTS
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64145885
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// @see https://github.com/ossrs/srs/issues/212#issuecomment-64145885
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// byte_alignment()
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// adts_fixed_header:
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@ -2260,7 +2260,7 @@ class SrsRawAacStream
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var channel_configuration:uint = (sfiv >> 6) & 0x07;
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/*int8_t original = (sfiv >> 5) & 0x01;*/
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/*int8_t home = (sfiv >> 4) & 0x01;*/
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//int8_t Emphasis; @remark, Emphasis is removed, @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64154736
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//int8_t Emphasis; @remark, Emphasis is removed, @see https://github.com/ossrs/srs/issues/212#issuecomment-64154736
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// 4bits left.
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// adts_variable_header(), 1.A.2.2.2 Variable Header of ADTS
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// copyright_identification_bit 1 bslbf
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@ -2357,7 +2357,7 @@ class SrsRawAacStream
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var samplingFrequencyIndex:uint = codec.sampling_frequency_index;
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// override the aac samplerate by user specified.
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64146899
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// @see https://github.com/ossrs/srs/issues/212#issuecomment-64146899
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switch (codec.sound_rate) {
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case SrsConsts.SrsCodecAudioSampleRate11025:
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samplingFrequencyIndex = 0x0a; break;
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@ -2478,7 +2478,7 @@ class SrsEnum
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/**
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* the aac profile, for ADTS(HLS/TS)
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* @see https://github.com/winlinvip/simple-rtmp-server/issues/310
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* @see https://github.com/ossrs/srs/issues/310
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*/
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class SrsAacProfile extends SrsEnum
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{
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@ -3697,7 +3697,7 @@ class SrsTsAdaptationField
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// @remark, for as, should never shift the Number object.
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// @remark, use pcr base and ignore the extension
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/250#issuecomment-71349370
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// @see https://github.com/ossrs/srs/issues/250#issuecomment-71349370
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// first 33bits is pcr base.
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program_clock_reference_base = (stream.readUnsignedInt() << 1) & 0x1fffffe;
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var ch:uint = stream.readUnsignedByte();
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@ -3715,7 +3715,7 @@ class SrsTsAdaptationField
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}
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// @remark, for as, should never shift the Number object.
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// @remark, use pcr base and ignore the extension
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/250#issuecomment-71349370
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// @see https://github.com/ossrs/srs/issues/250#issuecomment-71349370
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// first 33bits is pcr base.
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original_program_clock_reference_base = (stream.readUnsignedInt() << 1) & 0x1fffffe;
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ch = stream.readUnsignedByte();
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@ -45,7 +45,7 @@ function remote_check()
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ok_msg "remote $remote ok, url is $url"
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}
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remote_check origin git@github.com:ossrs/srs.git
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remote_check srs.winlin git@github.com:winlinvip/simple-rtmp-server.git
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remote_check srs.winlin git@github.com:winlinvip/srs.git
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remote_check srs.csdn git@code.csdn.net:winlinvip/srs-csdn.git
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remote_check srs.oschina git@git.oschina.net:winlinvip/srs.oschina.git
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remote_check srs.gitlab git@gitlab.com:winlinvip/srs-gitlab.git
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