Restruct by Ross Finlayson <finlayson@live.com>

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.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9251 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
bertrand 2003-02-03 10:27:50 +00:00
parent cbe52ff6e4
commit 06d22fab96
4 changed files with 341 additions and 164 deletions

View File

@ -10,7 +10,7 @@ endif
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

View File

@ -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 @@ typedef struct RTPState {
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 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
}
// 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 @@ extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
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 @@ extern "C" void demux_close_rtp(demuxer_t* demuxer) {
////////// 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 @@ static void afterReading(void* clientData, unsigned frameSize,
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
@ -458,10 +366,40 @@ static void onSourceClosure(void* clientData) {
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:
#define MAX_QUEUE_SIZE 5

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@ -0,0 +1,203 @@
////////// 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;
}

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@ -0,0 +1,36 @@
#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