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mpv/libmpdemux/demux_rtp.cpp

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extern "C" {
#include "demux_rtp.h"
#include "stheader.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;
stream_t* newStream = NULL;
do {
char* sdpDescription = (char*)malloc(fileSize+1);
if (sdpDescription == NULL) break;
ssize_t numBytesRead = read(fd, sdpDescription, fileSize);
if (numBytesRead != fileSize) break;
sdpDescription[fileSize] = '\0'; // to be safe
newStream = (stream_t*)calloc(sizeof (stream_t), 1);
if (newStream == NULL) break;
// Store the SDP description in the 'priv' field, for later use:
newStream->priv = sdpDescription;
} while (0);
return newStream;
}
extern "C" int _rtsp_streaming_seek(int /*fd*/, off_t /*pos*/,
streaming_ctrl_t* /*streaming_ctrl*/) {
return -1; // For now, we don't handle RTSP stream seeking
}
extern "C" int rtsp_streaming_start(stream_t* stream) {
stream->streaming_ctrl->streaming_seek = _rtsp_streaming_seek;
return 0;
}
// A data structure representing a buffer being read:
class ReadBufferQueue; // forward
class ReadBuffer {
public:
ReadBuffer(ReadBufferQueue* ourQueue, demux_packet_t* dp);
virtual ~ReadBuffer();
Boolean enqueue();
demux_packet_t* dp() const { return fDP; }
ReadBufferQueue* ourQueue() { return fOurQueue; }
ReadBuffer* next;
private:
demux_packet_t* fDP;
ReadBufferQueue* fOurQueue;
};
class ReadBufferQueue {
public:
ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
char const* tag);
virtual ~ReadBufferQueue();
ReadBuffer* dequeue();
FramedSource* readSource() const { return fReadSource; }
RTPSource* rtpSource() const { return fRTPSource; }
demuxer_t* ourDemuxer() const { return fOurDemuxer; }
char const* tag() const { return fTag; }
ReadBuffer* head;
ReadBuffer* tail;
char blockingFlag; // used to implement synchronous reads
unsigned counter; // used for debugging
private:
FramedSource* fReadSource;
RTPSource* fRTPSource;
demuxer_t* fOurDemuxer;
char const* fTag; // used for debugging
};
// A structure of RTP-specific state, kept so that we can cleanly
// reclaim it:
typedef struct RTPState {
char const* sdpDescription;
RTSPClient* rtspClient;
MediaSession* mediaSession;
ReadBufferQueue* audioBufferQueue;
ReadBufferQueue* videoBufferQueue;
int isMPEG; // TRUE for MPEG audio, video, or transport streams
struct timeval firstSyncTime;
};
int rtspStreamOverTCP = 0;
extern "C" void demux_open_rtp(demuxer_t* demuxer) {
if (rtspStreamOverTCP && LIVEMEDIA_LIBRARY_VERSION_INT < 1033689600) {
fprintf(stderr, "TCP streaming of RTP/RTCP requires \"LIVE.COM Streaming Media\" library version 2002.10.04 or later - ignoring the \"-rtsp-stream-over-tcp\" flag\n");
rtspStreamOverTCP = 0;
}
do {
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
if (scheduler == NULL) break;
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
if (env == NULL) break;
RTSPClient* rtspClient = NULL;
int isMPEG = 0;
// Look at the stream's 'priv' field to see if we were initiated
// via a SDP description:
char* sdpDescription = (char*)(demuxer->stream->priv);
if (sdpDescription == NULL) {
// We weren't given a SDP description directly, so assume that
// we were give a RTSP URL
char const* url = demuxer->stream->streaming_ctrl->url->url;
extern int verbose;
rtspClient = RTSPClient::createNew(*env, verbose, "mplayer");
if (rtspClient == NULL) {
fprintf(stderr, "Failed to create RTSP client: %s\n",
env->getResultMsg());
break;
}
sdpDescription = rtspClient->describeURL(url);
if (sdpDescription == NULL) {
fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
url, env->getResultMsg());
break;
}
}
// Now that we have a SDP description, create a MediaSession from it:
MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
if (mediaSession == NULL) break;
// Create RTP receivers (sources) for each subsession:
MediaSubsessionIterator iter(*mediaSession);
MediaSubsession* subsession;
MediaSubsession* audioSubsession = NULL;
MediaSubsession* videoSubsession = NULL;
while ((subsession = iter.next()) != NULL) {
// Ignore any subsession that's not audio or video:
if (strcmp(subsession->mediumName(), "audio") == 0) {
audioSubsession = subsession;
} else if (strcmp(subsession->mediumName(), "video") == 0) {
videoSubsession = subsession;
} else {
continue;
}
if (!subsession->initiate()) {
fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
} else {
fprintf(stderr, "Initiated \"%s/%s\" RTP subsession\n", subsession->mediumName(), subsession->codecName());
if (rtspClient != NULL) {
// Issue RTSP "SETUP" and "PLAY" commands on the chosen subsession:
if (!rtspClient->setupMediaSubsession(*subsession, False,
rtspStreamOverTCP)) break;
if (!rtspClient->playMediaSubsession(*subsession)) break;
}
// 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 {
fprintf(stderr,
"Unknown mplayer format code for MIME type \"audio/%s\"\n",
subsession->codecName());
}
}
}
}
// 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;
} 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;
}
static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue,
demux_stream_t* ds); // 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.
// Note that this is called as a synchronous read operation, so it needs
// to block in the (hopefully infrequent) case where no packet is
// immediately available.
// 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 (ds == demuxer->video) {
bufferQueue = rtpState->videoBufferQueue;
} else if (ds == demuxer->audio) {
bufferQueue = rtpState->audioBufferQueue;
} else {
fprintf(stderr, "demux_rtp_fill_buffer: internal error: unknown stream\n");
return 0;
}
if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
fprintf(stderr, "demux_rtp_fill_buffer failed: no appropriate RTP subsession has been set up\n");
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();
scheduler.blockMyself(&bufferQueue->blockingFlag);
}
if (demuxer->stream->eof) return 0; // source stream has closed down
return 1;
}
extern "C" void demux_close_rtp(demuxer_t* demuxer) {
// Reclaim all RTP-related state:
// Get the RTP state that was stored in the demuxer's 'priv' field:
RTPState* rtpState = (RTPState*)(demuxer->priv);
if (rtpState == NULL) return;
UsageEnvironment* env = NULL;
TaskScheduler* scheduler = NULL;
if (rtpState->mediaSession != NULL) {
env = &(rtpState->mediaSession->envir());
scheduler = &(env->taskScheduler());
}
Medium::close(rtpState->mediaSession);
Medium::close(rtpState->rtspClient);
delete rtpState->audioBufferQueue;
delete rtpState->videoBufferQueue;
delete rtpState->sdpDescription;
delete rtpState;
delete env; delete scheduler;
}
////////// 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
static void scheduleNewBufferRead(ReadBufferQueue* bufferQueue) {
if (bufferQueue->readSource()->isCurrentlyAwaitingData()) return;
// a read from this source is already in progress
// Allocate a new packet buffer, and arrange to read into it:
unsigned const bufferSize = 30000; // >= the largest conceivable RTP packet
demux_packet_t* dp = new_demux_packet(bufferSize);
if (dp == NULL) return;
ReadBuffer* readBuffer = new ReadBuffer(bufferQueue, dp);
// Schedule the read operation:
bufferQueue->readSource()->getNextFrame(dp->buffer, bufferSize,
afterReading, readBuffer,
onSourceClosure, readBuffer);
}
static void afterReading(void* clientData, unsigned frameSize,
struct timeval presentationTime) {
ReadBuffer* readBuffer = (ReadBuffer*)clientData;
ReadBufferQueue* bufferQueue = readBuffer->ourQueue();
demuxer_t* demuxer = bufferQueue->ourDemuxer();
RTPState* rtpState = (RTPState*)(demuxer->priv);
if (frameSize > 0) demuxer->stream->eof = 0;
demux_packet_t* dp = readBuffer->dp();
dp->len = frameSize;
// Set the packet's presentation time stamp, depending on whether or
// not our RTP source's timestamps have been synchronized yet:
{
Boolean hasBeenSynchronized
= bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
if (hasBeenSynchronized) {
struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
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:
dp->pts = presentationTime.tv_sec - fst->tv_sec
+ (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
} else {
dp->pts = 0.0;
}
}
dp->pos = demuxer->filepos;
demuxer->filepos += frameSize;
if (!readBuffer->enqueue()) {
// The queue is full, so discard the buffer:
delete readBuffer;
}
// Signal any pending 'blockMyself()' call on this queue:
bufferQueue->blockingFlag = ~0;
// Finally, arrange to do another read, if appropriate
scheduleNewBufferRead(bufferQueue);
}
static void onSourceClosure(void* clientData) {
ReadBuffer* readBuffer = (ReadBuffer*)clientData;
ReadBufferQueue* bufferQueue = readBuffer->ourQueue();
demuxer_t* demuxer = bufferQueue->ourDemuxer();
demuxer->stream->eof = 1;
// Signal any pending 'blockMyself()' call on this queue:
bufferQueue->blockingFlag = ~0;
}
////////// "ReadBuffer" and "ReadBufferQueue" implementation:
#define MAX_QUEUE_SIZE 5
ReadBuffer::ReadBuffer(ReadBufferQueue* ourQueue, demux_packet_t* dp)
: next(NULL), fDP(dp), fOurQueue(ourQueue) {
}
Boolean ReadBuffer::enqueue() {
if (fOurQueue->counter >= MAX_QUEUE_SIZE) {
// This queue is full. Clear out an old entry from it, so that
// this new one will fit:
while (fOurQueue->counter >= MAX_QUEUE_SIZE) {
delete fOurQueue->dequeue();
}
}
// Add ourselves to the tail of our queue:
if (fOurQueue->tail == NULL) {
fOurQueue->head = this;
} else {
fOurQueue->tail->next = this;
}
fOurQueue->tail = this;
++fOurQueue->counter;
return True;
}
ReadBuffer::~ReadBuffer() {
free_demux_packet(fDP);
delete next;
}
ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
demuxer_t* demuxer, char const* tag)
: head(NULL), tail(NULL), counter(0),
fReadSource(subsession == NULL ? NULL : subsession->readSource()),
fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
fOurDemuxer(demuxer), fTag(strdup(tag)) {
}
ReadBufferQueue::~ReadBufferQueue() {
delete head;
delete fTag;
}
ReadBuffer* ReadBufferQueue::dequeue() {
ReadBuffer* readBuffer = head;
if (readBuffer != NULL) {
head = readBuffer->next;
if (head == NULL) tail = NULL;
--counter;
readBuffer->next = NULL;
}
return readBuffer;
}
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