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author | Uoti Urpala <uau@glyph.nonexistent.invalid> | 2009-07-07 02:26:13 +0300 |
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committer | Uoti Urpala <uau@glyph.nonexistent.invalid> | 2009-07-07 02:34:35 +0300 |
commit | 0eb321bf2c1cc0e048faff26a01f86cdd3ec254f (patch) | |
tree | 71cb9bd9ed121156d3382066c0722c73189afe04 /libmpdemux/demux_rtp.cpp | |
parent | 6d908205fbadbdf7ccdc6c5e0eb918f0b43f16e0 (diff) | |
download | mpv-0eb321bf2c1cc0e048faff26a01f86cdd3ec254f.tar.bz2 mpv-0eb321bf2c1cc0e048faff26a01f86cdd3ec254f.tar.xz |
Remove trailing whitespace from most files
Diffstat (limited to 'libmpdemux/demux_rtp.cpp')
-rw-r--r-- | libmpdemux/demux_rtp.cpp | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp index 1cc6a7c868..fe163aaf5a 100644 --- a/libmpdemux/demux_rtp.cpp +++ b/libmpdemux/demux_rtp.cpp @@ -92,7 +92,7 @@ typedef struct RTPState { extern "C" char* network_username; extern "C" char* network_password; static char* openURL_rtsp(RTSPClient* client, char const* url) { - // If we were given a user name (and optional password), then use them: + // If we were given a user name (and optional password), then use them: if (network_username != NULL) { char const* password = network_password == NULL ? "" : network_password; return client->describeWithPassword(url, network_username, password); @@ -102,7 +102,7 @@ static char* openURL_rtsp(RTSPClient* client, char const* url) { } static char* openURL_sip(SIPClient* client, char const* url) { - // If we were given a user name (and optional password), then use them: + // If we were given a user name (and optional password), then use them: if (network_username != NULL) { char const* password = network_password == NULL ? "" : network_password; return client->inviteWithPassword(url, network_username, password); @@ -111,7 +111,7 @@ static char* openURL_sip(SIPClient* client, char const* url) { } } -int rtspStreamOverTCP = 0; +int rtspStreamOverTCP = 0; extern int rtsp_port; extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { @@ -127,7 +127,7 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { SIPClient* sipClient = NULL; if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen - demuxer->stream->eof = 0; // just in case + demuxer->stream->eof = 0; // just in case // Look at the stream's 'priv' field to see if we were initiated // via a SDP description: @@ -208,7 +208,7 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { if (rtsp_port) subsession->setClientPortNum (rtsp_port); - + if (!subsession->initiate()) { fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg()); } else { @@ -314,7 +314,7 @@ extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { if (dp == NULL) return 0; if (demuxer->stream->eof) return 0; // source stream has closed down - + // Before using this packet, check to make sure that its presentation // time is not far behind the other stream (if any). If it is, // then we discard this packet, and get another instead. (The rest of @@ -333,7 +333,7 @@ extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { ds_add_packet(ds, dp); break; } - + #ifdef DEBUG_PRINT_DISCARDED_PACKETS RTPState* rtpState = (RTPState*)(demuxer->priv); ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue; @@ -424,7 +424,7 @@ static void afterReading(void* clientData, unsigned frameSize, resize_demux_packet(dp, frameSize + headersize); // Set the packet's presentation time stamp, depending on whether or - // not our RTP source's timestamps have been synchronized yet: + // not our RTP source's timestamps have been synchronized yet: Boolean hasBeenSynchronized = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP(); if (hasBeenSynchronized) { @@ -437,7 +437,7 @@ static void afterReading(void* clientData, unsigned frameSize, if (fst->tv_sec == 0 && fst->tv_usec == 0) { *fst = presentationTime; } - + // For the "pts" field, use the time differential from the first // synchronized time, rather than absolute time, in order to avoid // round-off errors when converting to a float: @@ -502,7 +502,7 @@ static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n"); return NULL; } - + demux_packet_t* dp = NULL; if (!mustGetNewData) { // Check whether we have a previously-saved buffer that we can use: @@ -619,7 +619,7 @@ ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, fReadSource(subsession == NULL ? NULL : subsession->readSource()), fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()), fOurDemuxer(demuxer), fTag(strdup(tag)) { -} +} ReadBufferQueue::~ReadBufferQueue() { delete fTag; @@ -649,7 +649,7 @@ demux_packet_t* ReadBufferQueue::getPendingBuffer() { demux_packet_t* dp = pendingDPHead; if (dp != NULL) { pendingDPHead = dp->next; - if (pendingDPHead == NULL) pendingDPTail = NULL; + if (pendingDPHead == NULL) pendingDPTail = NULL; dp->next = NULL; } |