[MPlayer-dev-eng] [PATCH] Cleanup to the LIVE.COM interface code, & more QT stream support
Ross Finlayson
finlayson at live.com
Fri Jan 31 03:38:07 CET 2003
I have made some improvements to the LIVE.COM interface code. The code now
supports 'QuickTime generic' RTP streams (the "X-QT" MIME type), which -
thanks to the QuickTime codecs - makes it possible to play more QuickTime
RTP streams. ("X-SORENSONVIDEO" RTP streams are still not supported,
though, because I haven't yet reverse-engineered that RTP payload format.)
QCELP RTP streams are also supported (although they don't play quite right
just yet).
I have also restructured the code, splitting it into two files:
- "demux_rtp.cpp" (the existing file), which handles RTP I/O and
MPlayer buffering. This file should require few changes in the future.
- "demux_rtp_codec.cpp" (a new file), which does codec-specific
initialization for each media type. In the future, most changes (to
support new media types) will be to this file only.
There's also a new header file
- "demux_rtp_internal.h", which ties together the two files
mentioned above.
I have included a patch to the existing files "libmpdemux/demux_rtp.cpp"
and "libmpdemux/Makefile", plus the two new files
"libmpdemux/demux_rtp_codec.cpp" and "libmpdemux/demux_rtp_internal.h".
Ross.
-------------- next part --------------
Index: libmpdemux/Makefile
===================================================================
RCS file: /cvsroot/mplayer/main/libmpdemux/Makefile,v
retrieving revision 1.49
diff -u -r1.49 Makefile
--- libmpdemux/Makefile 22 Jan 2003 23:51:04 -0000 1.49
+++ libmpdemux/Makefile 30 Jan 2003 11:13:36 -0000
@@ -10,7 +10,7 @@
ifeq ($(STREAMING),yes)
SRCS += asf_streaming.c url.c http.c network.c asf_mmst_streaming.c pnm.c
ifeq ($(STREAMING_LIVE_DOT_COM),yes)
-CPLUSPLUSSRCS = demux_rtp.cpp
+CPLUSPLUSSRCS = demux_rtp.cpp demux_rtp_codec.cpp
CPLUSPLUSINCLUDE = -I$(LIVE_LIB_DIR)/liveMedia/include
CPLUSPLUSINCLUDE += -I$(LIVE_LIB_DIR)/UsageEnvironment/include
CPLUSPLUSINCLUDE += -I$(LIVE_LIB_DIR)/BasicUsageEnvironment/include
Index: libmpdemux/demux_rtp.cpp
===================================================================
RCS file: /cvsroot/mplayer/main/libmpdemux/demux_rtp.cpp,v
retrieving revision 1.9
diff -u -r1.9 demux_rtp.cpp
--- libmpdemux/demux_rtp.cpp 18 Jan 2003 14:22:30 -0000 1.9
+++ libmpdemux/demux_rtp.cpp 30 Jan 2003 11:13:37 -0000
@@ -1,15 +1,16 @@
+////////// Routines (with C-linkage) that interface between "MPlayer"
+////////// and the "LIVE.COM Streaming Media" libraries:
+
extern "C" {
#include "demux_rtp.h"
#include "stheader.h"
}
+#include "demux_rtp_internal.h"
#include "BasicUsageEnvironment.hh"
#include "liveMedia.hh"
#include <unistd.h>
-////////// Routines (with C-linkage) that interface between "mplayer"
-////////// and the "LIVE.COM Streaming Media" libraries:
-
extern "C" stream_t* stream_open_sdp(int fd, off_t fileSize,
int* file_format) {
*file_format = DEMUXER_TYPE_RTP;
@@ -91,7 +92,7 @@
MediaSession* mediaSession;
ReadBufferQueue* audioBufferQueue;
ReadBufferQueue* videoBufferQueue;
- int isMPEG; // TRUE for MPEG audio, video, or transport streams
+ unsigned flags;
struct timeval firstSyncTime;
};
@@ -109,7 +110,7 @@
if (env == NULL) break;
RTSPClient* rtspClient = NULL;
- int isMPEG = 0;
+ unsigned flags = 0;
// Look at the stream's 'priv' field to see if we were initiated
// via a SDP description:
@@ -120,7 +121,7 @@
char const* url = demuxer->stream->streaming_ctrl->url->url;
extern int verbose;
- rtspClient = RTSPClient::createNew(*env, verbose, "mplayer");
+ rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
if (rtspClient == NULL) {
fprintf(stderr, "Failed to create RTSP client: %s\n",
env->getResultMsg());
@@ -139,17 +140,26 @@
MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
if (mediaSession == NULL) break;
+
+ // Create a 'RTPState' structure containing the state that we just created,
+ // and store it in the demuxer's 'priv' field, for future reference:
+ RTPState* rtpState = new RTPState;
+ rtpState->sdpDescription = sdpDescription;
+ rtpState->rtspClient = rtspClient;
+ rtpState->mediaSession = mediaSession;
+ rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
+ demuxer->priv = rtpState;
+
// Create RTP receivers (sources) for each subsession:
MediaSubsessionIterator iter(*mediaSession);
MediaSubsession* subsession;
- MediaSubsession* audioSubsession = NULL;
- MediaSubsession* videoSubsession = NULL;
+ unsigned streamType = 0; // 0 => video; 1 => audio
while ((subsession = iter.next()) != NULL) {
// Ignore any subsession that's not audio or video:
if (strcmp(subsession->mediumName(), "audio") == 0) {
- audioSubsession = subsession;
+ streamType = 1;
} else if (strcmp(subsession->mediumName(), "video") == 0) {
- videoSubsession = subsession;
+ streamType = 0;
} else {
continue;
}
@@ -167,137 +177,31 @@
}
// Now that the subsession is ready to be read, do additional
- // mplayer-specific initialization on it:
- if (subsession == videoSubsession) {
- // Create a dummy video stream header
- // to make the main mplayer code happy:
- sh_video_t* sh_video = new_sh_video(demuxer,0);
- BITMAPINFOHEADER* bih
- = (BITMAPINFOHEADER*)calloc(1,sizeof(BITMAPINFOHEADER));
- bih->biSize = sizeof(BITMAPINFOHEADER);
- sh_video->bih = bih;
- demux_stream_t* d_video = demuxer->video;
- d_video->sh = sh_video; sh_video->ds = d_video;
-
- // If we happen to know the subsession's video frame rate, set it,
- // so that the user doesn't have to give the "-fps" option instead.
- int fps = (int)(subsession->videoFPS());
- if (fps != 0) sh_video->fps = fps;
-
- // Map known video MIME types to the BITMAPINFOHEADER parameters
- // that this program uses. (Note that not all types need all
- // of the parameters to be set.)
- if (strcmp(subsession->codecName(), "MPV") == 0 ||
- strcmp(subsession->codecName(), "MP1S") == 0 ||
- strcmp(subsession->codecName(), "MP2T") == 0) {
- isMPEG = 1;
- } else if (strcmp(subsession->codecName(), "H263") == 0 ||
- strcmp(subsession->codecName(), "H263-1998") == 0) {
- bih->biCompression = sh_video->format
- = mmioFOURCC('H','2','6','3');
- } else if (strcmp(subsession->codecName(), "H261") == 0) {
- bih->biCompression = sh_video->format
- = mmioFOURCC('H','2','6','1');
- } else {
- fprintf(stderr,
- "Unknown mplayer format code for MIME type \"video/%s\"\n",
- subsession->codecName());
- }
- } else if (subsession == audioSubsession) {
- // Create a dummy audio stream header
- // to make the main mplayer code happy:
- sh_audio_t* sh_audio = new_sh_audio(demuxer,0);
- WAVEFORMATEX* wf = (WAVEFORMATEX*)calloc(1,sizeof(WAVEFORMATEX));
- sh_audio->wf = wf;
- demux_stream_t* d_audio = demuxer->audio;
- d_audio->sh = sh_audio; sh_audio->ds = d_audio;
-
- // Map known audio MIME types to the WAVEFORMATEX parameters
- // that this program uses. (Note that not all types need all
- // of the parameters to be set.)
- wf->nSamplesPerSec
- = subsession->rtpSource()->timestampFrequency(); // by default
- if (strcmp(subsession->codecName(), "MPA") == 0 ||
- strcmp(subsession->codecName(), "MPA-ROBUST") == 0 ||
- strcmp(subsession->codecName(), "X-MP3-DRAFT-00") == 0) {
- wf->wFormatTag = sh_audio->format = 0x55;
- // Note: 0x55 is for layer III, but should work for I,II also
- wf->nSamplesPerSec = 0; // sample rate is deduced from the data
- } else if (strcmp(subsession->codecName(), "AC3") == 0) {
- wf->wFormatTag = sh_audio->format = 0x2000;
- wf->nSamplesPerSec = 0; // sample rate is deduced from the data
- } else if (strcmp(subsession->codecName(), "PCMU") == 0) {
- wf->wFormatTag = sh_audio->format = 0x7;
- wf->nChannels = 1;
- wf->nAvgBytesPerSec = 8000;
- wf->nBlockAlign = 1;
- wf->wBitsPerSample = 8;
- wf->cbSize = 0;
- } else if (strcmp(subsession->codecName(), "PCMA") == 0) {
- wf->wFormatTag = sh_audio->format = 0x6;
- wf->nChannels = 1;
- wf->nAvgBytesPerSec = 8000;
- wf->nBlockAlign = 1;
- wf->wBitsPerSample = 8;
- wf->cbSize = 0;
- } else if (strcmp(subsession->codecName(), "GSM") == 0) {
- wf->wFormatTag = sh_audio->format = mmioFOURCC('a','g','s','m');
- wf->nChannels = 1;
- wf->nAvgBytesPerSec = 1650;
- wf->nBlockAlign = 33;
- wf->wBitsPerSample = 16;
- wf->cbSize = 0;
- } else if (strcmp(subsession->codecName(), "MP4A-LATM") == 0) {
- wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
-#ifndef HAVE_FAAD
- fprintf(stderr, "WARNING: Playing MPEG-4 (AAC) Audio requires the \"faad\" library!\n");
-#endif
-#if (LIVEMEDIA_LIBRARY_VERSION_INT < 1042761600)
- fprintf(stderr, "WARNING: This audio stream might not play correctly. Please upgrade to version \"2003.01.17\" or later of the \"LIVE.COM Streaming Media\" libraries.\n");
-#else
- // For the codec to work correctly, it needs "AudioSpecificConfig"
- // data, which is parsed from the "StreamMuxConfig" string that
- // was present (hopefully) in the SDP description:
- unsigned codecdata_len;
- sh_audio->codecdata
- = parseStreamMuxConfigStr(subsession->fmtp_config(),
- codecdata_len);
- sh_audio->codecdata_len = codecdata_len;
-#endif
- } else {
- fprintf(stderr,
- "Unknown mplayer format code for MIME type \"audio/%s\"\n",
- subsession->codecName());
- }
+ // MPlayer codec-specific initialization on it:
+ if (streamType == 0) { // video
+ rtpState->videoBufferQueue
+ = new ReadBufferQueue(subsession, demuxer, "video");
+ rtpCodecInitialize_video(demuxer, subsession, flags);
+ } else { // audio
+ rtpState->audioBufferQueue
+ = new ReadBufferQueue(subsession, demuxer, "audio");
+ rtpCodecInitialize_audio(demuxer, subsession, flags);
}
}
}
-
- // Hack: Create a 'RTPState' structure containing the state that
- // we just created, and store it in the demuxer's 'priv' field:
- RTPState* rtpState = new RTPState;
- rtpState->sdpDescription = sdpDescription;
- rtpState->rtspClient = rtspClient;
- rtpState->mediaSession = mediaSession;
- rtpState->audioBufferQueue
- = new ReadBufferQueue(audioSubsession, demuxer, "audio");
- rtpState->videoBufferQueue
- = new ReadBufferQueue(videoSubsession, demuxer, "video");
- rtpState->isMPEG = isMPEG;
- rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
-
- demuxer->priv = rtpState;
+ rtpState->flags = flags;
} while (0);
}
extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
// Get the RTP state that was stored in the demuxer's 'priv' field:
RTPState* rtpState = (RTPState*)(demuxer->priv);
- return rtpState->isMPEG;
+
+ return (rtpState->flags&RTPSTATE_IS_MPEG) != 0;
}
-static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue,
- demux_stream_t* ds); // forward
+static ReadBuffer* getBuffer(ReadBufferQueue* bufferQueue,
+ demuxer_t* demuxer); // forward
extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
// Get a filled-in "demux_packet" from the RTP source, and deliver it.
@@ -324,24 +228,46 @@
return 0;
}
- // Check whether there's a full buffer to deliver to the client:
- bufferQueue->blockingFlag = 0;
- while (!deliverBufferIfAvailable(bufferQueue, ds)) {
- // Because we weren't able to deliver a buffer to the client immediately,
- // block myself until one comes available:
- TaskScheduler& scheduler
- = bufferQueue->readSource()->envir().taskScheduler();
-#if USAGEENVIRONMENT_LIBRARY_VERSION_INT >= 1038614400
- scheduler.doEventLoop(&bufferQueue->blockingFlag);
-#else
- scheduler.blockMyself(&bufferQueue->blockingFlag);
-#endif
- }
+ ReadBuffer* readBuffer = getBuffer(bufferQueue, demuxer); // blocking
+ if (readBuffer != NULL) ds_add_packet(ds, readBuffer->dp());
if (demuxer->stream->eof) return 0; // source stream has closed down
+
return 1;
}
+Boolean awaitRTPPacket(demuxer_t* demuxer, unsigned streamType,
+ unsigned char*& packetData, unsigned& packetDataLen) {
+ // Begin by finding the buffer queue that we want to read from:
+ // (Get this from the RTP state, which we stored in
+ // the demuxer's 'priv' field)
+ RTPState* rtpState = (RTPState*)(demuxer->priv);
+ ReadBufferQueue* bufferQueue = NULL;
+ if (streamType == 0) {
+ bufferQueue = rtpState->videoBufferQueue;
+ } else if (streamType == 1) {
+ bufferQueue = rtpState->audioBufferQueue;
+ } else {
+ fprintf(stderr, "awaitRTPPacket: internal error: unknown streamType %d\n",
+ streamType);
+ return False;
+ }
+
+ if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
+ fprintf(stderr, "awaitRTPPacket failed: no appropriate RTP subsession has been set up\n");
+ return False;
+ }
+
+ ReadBuffer* readBuffer = getBuffer(bufferQueue, demuxer); // blocking
+ if (readBuffer == NULL) return False;
+
+ demux_packet_t* dp = readBuffer->dp();
+ packetData = dp->buffer;
+ packetDataLen = dp->len;
+
+ return True;
+}
+
extern "C" void demux_close_rtp(demuxer_t* demuxer) {
// Reclaim all RTP-related state:
@@ -366,24 +292,6 @@
////////// Extra routines that help implement the above interface functions:
-static void scheduleNewBufferRead(ReadBufferQueue* bufferQueue); // forward
-
-static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue,
- demux_stream_t* ds) {
- Boolean deliveredBuffer = False;
- ReadBuffer* readBuffer = bufferQueue->dequeue();
- if (readBuffer != NULL) {
- // Append the packet to the reader's DS stream:
- ds_add_packet(ds, readBuffer->dp());
- deliveredBuffer = True;
- }
-
- // Arrange to read a new packet into this queue:
- scheduleNewBufferRead(bufferQueue);
-
- return deliveredBuffer;
-}
-
static void afterReading(void* clientData, unsigned frameSize,
struct timeval presentationTime); // forward
static void onSourceClosure(void* clientData); // forward
@@ -444,7 +352,7 @@
delete readBuffer;
}
- // Signal any pending 'blockMyself()' call on this queue:
+ // Signal any pending 'doEventLoop()' call on this queue:
bufferQueue->blockingFlag = ~0;
// Finally, arrange to do another read, if appropriate
@@ -458,8 +366,38 @@
demuxer->stream->eof = 1;
- // Signal any pending 'blockMyself()' call on this queue:
+ // Signal any pending 'doEventLoop()' call on this queue:
bufferQueue->blockingFlag = ~0;
+}
+
+static ReadBuffer* getBufferIfAvailable(ReadBufferQueue* bufferQueue) {
+ ReadBuffer* readBuffer = bufferQueue->dequeue();
+
+ // Arrange to read a new packet into this queue:
+ scheduleNewBufferRead(bufferQueue);
+
+ return readBuffer;
+}
+
+static ReadBuffer* getBuffer(ReadBufferQueue* bufferQueue,
+ demuxer_t* demuxer) {
+ // Check whether there's a full buffer to deliver to the client:
+ bufferQueue->blockingFlag = 0;
+ ReadBuffer* readBuffer;
+ while ((readBuffer = getBufferIfAvailable(bufferQueue)) == NULL
+ && !demuxer->stream->eof) {
+ // Because we weren't able to deliver a buffer to the client immediately,
+ // block myself until one comes available:
+ TaskScheduler& scheduler
+ = bufferQueue->readSource()->envir().taskScheduler();
+#if USAGEENVIRONMENT_LIBRARY_VERSION_INT >= 1038614400
+ scheduler.doEventLoop(&bufferQueue->blockingFlag);
+#else
+ scheduler.blockMyself(&bufferQueue->blockingFlag);
+#endif
+ }
+
+ return readBuffer;
}
////////// "ReadBuffer" and "ReadBufferQueue" implementation:
-------------- next part --------------
////////// Codec-specific routines used to interface between "MPlayer"
////////// and the "LIVE.COM Streaming Media" libraries:
#include "demux_rtp_internal.h"
extern "C" {
#include "stheader.h"
}
static Boolean
parseQTState_video(QuickTimeGenericRTPSource::QTState const& qtState,
unsigned& fourcc); // forward
static Boolean
parseQTState_audio(QuickTimeGenericRTPSource::QTState const& qtState,
unsigned& fourcc, unsigned& numChannels); // forward
void rtpCodecInitialize_video(demuxer_t* demuxer,
MediaSubsession* subsession,
unsigned& flags) {
flags = 0;
// Create a dummy video stream header
// to make the main MPlayer code happy:
sh_video_t* sh_video = new_sh_video(demuxer,0);
BITMAPINFOHEADER* bih
= (BITMAPINFOHEADER*)calloc(1,sizeof(BITMAPINFOHEADER));
bih->biSize = sizeof(BITMAPINFOHEADER);
sh_video->bih = bih;
demux_stream_t* d_video = demuxer->video;
d_video->sh = sh_video; sh_video->ds = d_video;
// If we happen to know the subsession's video frame rate, set it,
// so that the user doesn't have to give the "-fps" option instead.
int fps = (int)(subsession->videoFPS());
if (fps != 0) sh_video->fps = fps;
// Map known video MIME types to the BITMAPINFOHEADER parameters
// that this program uses. (Note that not all types need all
// of the parameters to be set.)
if (strcmp(subsession->codecName(), "MPV") == 0 ||
strcmp(subsession->codecName(), "MP1S") == 0 ||
strcmp(subsession->codecName(), "MP2T") == 0) {
flags |= RTPSTATE_IS_MPEG;
} else if (strcmp(subsession->codecName(), "H263") == 0 ||
strcmp(subsession->codecName(), "H263-1998") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('H','2','6','3');
} else if (strcmp(subsession->codecName(), "H261") == 0) {
bih->biCompression = sh_video->format
= mmioFOURCC('H','2','6','1');
} else if (strcmp(subsession->codecName(), "X-QT") == 0 ||
strcmp(subsession->codecName(), "X-QUICKTIME") == 0) {
// QuickTime generic RTP format, as described in
// http://developer.apple.com/quicktime/icefloe/dispatch026.html
// We can't initialize this stream until we've received the first packet
// that has QuickTime "sdAtom" information in the header. So, keep
// reading packets until we get one:
unsigned char* packetData; unsigned packetDataLen;
QuickTimeGenericRTPSource* qtRTPSource
= (QuickTimeGenericRTPSource*)(subsession->rtpSource());
unsigned fourcc;
do {
if (!awaitRTPPacket(demuxer, 0 /*video*/, packetData, packetDataLen)) {
return;
}
} while (!parseQTState_video(qtRTPSource->qtState, fourcc));
bih->biCompression = sh_video->format = fourcc;
} else {
fprintf(stderr,
"Unknown MPlayer format code for MIME type \"video/%s\"\n",
subsession->codecName());
}
}
void rtpCodecInitialize_audio(demuxer_t* demuxer,
MediaSubsession* subsession,
unsigned& flags) {
flags = 0;
// Create a dummy audio stream header
// to make the main MPlayer code happy:
sh_audio_t* sh_audio = new_sh_audio(demuxer,0);
WAVEFORMATEX* wf = (WAVEFORMATEX*)calloc(1,sizeof(WAVEFORMATEX));
sh_audio->wf = wf;
demux_stream_t* d_audio = demuxer->audio;
d_audio->sh = sh_audio; sh_audio->ds = d_audio;
// Map known audio MIME types to the WAVEFORMATEX parameters
// that this program uses. (Note that not all types need all
// of the parameters to be set.)
wf->nSamplesPerSec
= subsession->rtpSource()->timestampFrequency(); // by default
if (strcmp(subsession->codecName(), "MPA") == 0 ||
strcmp(subsession->codecName(), "MPA-ROBUST") == 0 ||
strcmp(subsession->codecName(), "X-MP3-DRAFT-00") == 0) {
wf->wFormatTag = sh_audio->format = 0x55;
// Note: 0x55 is for layer III, but should work for I,II also
wf->nSamplesPerSec = 0; // sample rate is deduced from the data
flags |= RTPSTATE_IS_MPEG;
} else if (strcmp(subsession->codecName(), "AC3") == 0) {
wf->wFormatTag = sh_audio->format = 0x2000;
wf->nSamplesPerSec = 0; // sample rate is deduced from the data
} else if (strcmp(subsession->codecName(), "PCMU") == 0) {
wf->wFormatTag = sh_audio->format = 0x7;
wf->nChannels = 1;
wf->nAvgBytesPerSec = 8000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "PCMA") == 0) {
wf->wFormatTag = sh_audio->format = 0x6;
wf->nChannels = 1;
wf->nAvgBytesPerSec = 8000;
wf->nBlockAlign = 1;
wf->wBitsPerSample = 8;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "GSM") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('a','g','s','m');
wf->nChannels = 1;
wf->nAvgBytesPerSec = 1650;
wf->nBlockAlign = 33;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "QCELP") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('Q','c','l','p');
// The following settings for QCELP don't quite work right #####
wf->nChannels = 1;
wf->nAvgBytesPerSec = 1750;
wf->nBlockAlign = 35;
wf->wBitsPerSample = 16;
wf->cbSize = 0;
} else if (strcmp(subsession->codecName(), "MP4A-LATM") == 0) {
wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
#ifndef HAVE_FAAD
fprintf(stderr, "WARNING: Playing MPEG-4 (AAC) Audio requires the \"faad\" library!\n");
#endif
#if (LIVEMEDIA_LIBRARY_VERSION_INT < 1042761600)
fprintf(stderr, "WARNING: This audio stream might not play correctly. Please upgrade to version \"2003.01.17\" or later of the \"LIVE.COM Streaming Media\" libraries.\n");
#else
// For the codec to work correctly, it needs "AudioSpecificConfig"
// data, which is parsed from the "StreamMuxConfig" string that
// was present (hopefully) in the SDP description:
unsigned codecdata_len;
sh_audio->codecdata
= parseStreamMuxConfigStr(subsession->fmtp_config(),
codecdata_len);
sh_audio->codecdata_len = codecdata_len;
#endif
flags |= RTPSTATE_IS_MPEG;
} else if (strcmp(subsession->codecName(), "X-QT") == 0 ||
strcmp(subsession->codecName(), "X-QUICKTIME") == 0) {
// QuickTime generic RTP format, as described in
// http://developer.apple.com/quicktime/icefloe/dispatch026.html
// We can't initialize this stream until we've received the first packet
// that has QuickTime "sdAtom" information in the header. So, keep
// reading packets until we get one:
unsigned char* packetData; unsigned packetDataLen;
QuickTimeGenericRTPSource* qtRTPSource
= (QuickTimeGenericRTPSource*)(subsession->rtpSource());
unsigned fourcc, numChannels;
do {
if (!awaitRTPPacket(demuxer, 1 /*audio*/, packetData, packetDataLen)) {
return;
}
} while (!parseQTState_audio(qtRTPSource->qtState, fourcc, numChannels));
wf->wFormatTag = sh_audio->format = fourcc;
wf->nChannels = numChannels;
} else {
fprintf(stderr,
"Unknown MPlayer format code for MIME type \"audio/%s\"\n",
subsession->codecName());
}
}
static Boolean
parseQTState_video(QuickTimeGenericRTPSource::QTState const& qtState,
unsigned& fourcc) {
// qtState's "sdAtom" field is supposed to contain a QuickTime video
// 'sample description' atom. This atom's name is the 'fourcc' that we want:
char const* sdAtom = qtState.sdAtom;
if (sdAtom == NULL || qtState.sdAtomSize < 2*4) return False;
fourcc = *(unsigned*)(&sdAtom[4]); // put in host order
return True;
}
static Boolean
parseQTState_audio(QuickTimeGenericRTPSource::QTState const& qtState,
unsigned& fourcc, unsigned& numChannels) {
// qtState's "sdAtom" field is supposed to contain a QuickTime audio
// 'sample description' atom. This atom's name is the 'fourcc' that we want.
// Also, the top half of the 5th word following the atom name should
// contain the number of channels ("numChannels") that we want:
char const* sdAtom = qtState.sdAtom;
if (sdAtom == NULL || qtState.sdAtomSize < 7*4) return False;
fourcc = *(unsigned*)(&sdAtom[4]); // put in host order
char const* word7Ptr = &sdAtom[6*4];
numChannels = (word7Ptr[0]<<8)|(word7Ptr[1]);
return True;
}
-------------- next part --------------
#ifndef _DEMUX_RTP_INTERNAL_H
#define _DEMUX_RTP_INTERNAL_H
#include <stdlib.h>
extern "C" {
#ifndef __STREAM_H
#include "stream.h"
#endif
#ifndef __DEMUXER_H
#include "demuxer.h"
#endif
}
#ifndef _LIVEMEDIA_HH
#include <liveMedia.hh>
#endif
// Codec-specific initialization routines:
void rtpCodecInitialize_video(demuxer_t* demuxer,
MediaSubsession* subsession, unsigned& flags);
void rtpCodecInitialize_audio(demuxer_t* demuxer,
MediaSubsession* subsession, unsigned& flags);
// Flags that may be set by the above routines:
#define RTPSTATE_IS_MPEG 0x1 // is an MPEG audio, video or transport stream
// A routine to wait for the first packet of a RTP stream to arrive.
// (For some RTP payload formats, codecs cannot be fully initialized until
// we've started receiving data.)
Boolean awaitRTPPacket(demuxer_t* demuxer, unsigned streamType,
unsigned char*& packetData, unsigned& packetDataLen);
// "streamType": 0 => video; 1 => audio
// This routine returns False if the input stream has closed
#endif
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