mirror of https://github.com/mpv-player/mpv
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:
parent
cbe52ff6e4
commit
06d22fab96
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@ -10,7 +10,7 @@ endif
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ifeq ($(STREAMING),yes)
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SRCS += asf_streaming.c url.c http.c network.c asf_mmst_streaming.c pnm.c
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ifeq ($(STREAMING_LIVE_DOT_COM),yes)
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CPLUSPLUSSRCS = demux_rtp.cpp
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CPLUSPLUSSRCS = demux_rtp.cpp demux_rtp_codec.cpp
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CPLUSPLUSINCLUDE = -I$(LIVE_LIB_DIR)/liveMedia/include
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CPLUSPLUSINCLUDE += -I$(LIVE_LIB_DIR)/UsageEnvironment/include
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CPLUSPLUSINCLUDE += -I$(LIVE_LIB_DIR)/BasicUsageEnvironment/include
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@ -1,15 +1,16 @@
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////////// Routines (with C-linkage) that interface between "MPlayer"
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////////// and the "LIVE.COM Streaming Media" libraries:
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extern "C" {
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#include "demux_rtp.h"
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#include "stheader.h"
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}
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#include "demux_rtp_internal.h"
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#include "BasicUsageEnvironment.hh"
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#include "liveMedia.hh"
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#include <unistd.h>
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////////// Routines (with C-linkage) that interface between "mplayer"
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////////// and the "LIVE.COM Streaming Media" libraries:
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extern "C" stream_t* stream_open_sdp(int fd, off_t fileSize,
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int* file_format) {
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*file_format = DEMUXER_TYPE_RTP;
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@ -91,7 +92,7 @@ typedef struct RTPState {
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MediaSession* mediaSession;
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ReadBufferQueue* audioBufferQueue;
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ReadBufferQueue* videoBufferQueue;
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int isMPEG; // TRUE for MPEG audio, video, or transport streams
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unsigned flags;
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struct timeval firstSyncTime;
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};
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@ -109,7 +110,7 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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if (env == NULL) break;
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RTSPClient* rtspClient = NULL;
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int isMPEG = 0;
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unsigned flags = 0;
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// Look at the stream's 'priv' field to see if we were initiated
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// via a SDP description:
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@ -120,7 +121,7 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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char const* url = demuxer->stream->streaming_ctrl->url->url;
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extern int verbose;
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rtspClient = RTSPClient::createNew(*env, verbose, "mplayer");
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rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
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if (rtspClient == NULL) {
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fprintf(stderr, "Failed to create RTSP client: %s\n",
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env->getResultMsg());
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@ -139,17 +140,26 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
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if (mediaSession == NULL) break;
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// Create a 'RTPState' structure containing the state that we just created,
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// and store it in the demuxer's 'priv' field, for future reference:
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RTPState* rtpState = new RTPState;
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rtpState->sdpDescription = sdpDescription;
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rtpState->rtspClient = rtspClient;
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rtpState->mediaSession = mediaSession;
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rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
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demuxer->priv = rtpState;
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// Create RTP receivers (sources) for each subsession:
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MediaSubsessionIterator iter(*mediaSession);
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MediaSubsession* subsession;
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MediaSubsession* audioSubsession = NULL;
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MediaSubsession* videoSubsession = NULL;
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unsigned streamType = 0; // 0 => video; 1 => audio
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while ((subsession = iter.next()) != NULL) {
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// Ignore any subsession that's not audio or video:
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if (strcmp(subsession->mediumName(), "audio") == 0) {
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audioSubsession = subsession;
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streamType = 1;
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} else if (strcmp(subsession->mediumName(), "video") == 0) {
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videoSubsession = subsession;
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streamType = 0;
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} else {
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continue;
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}
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@ -167,137 +177,31 @@ extern "C" void demux_open_rtp(demuxer_t* demuxer) {
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}
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// Now that the subsession is ready to be read, do additional
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// mplayer-specific initialization on it:
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if (subsession == videoSubsession) {
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// Create a dummy video stream header
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// to make the main mplayer code happy:
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sh_video_t* sh_video = new_sh_video(demuxer,0);
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BITMAPINFOHEADER* bih
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= (BITMAPINFOHEADER*)calloc(1,sizeof(BITMAPINFOHEADER));
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bih->biSize = sizeof(BITMAPINFOHEADER);
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sh_video->bih = bih;
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demux_stream_t* d_video = demuxer->video;
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d_video->sh = sh_video; sh_video->ds = d_video;
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// If we happen to know the subsession's video frame rate, set it,
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// so that the user doesn't have to give the "-fps" option instead.
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int fps = (int)(subsession->videoFPS());
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if (fps != 0) sh_video->fps = fps;
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// Map known video MIME types to the BITMAPINFOHEADER parameters
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// that this program uses. (Note that not all types need all
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// of the parameters to be set.)
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if (strcmp(subsession->codecName(), "MPV") == 0 ||
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strcmp(subsession->codecName(), "MP1S") == 0 ||
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strcmp(subsession->codecName(), "MP2T") == 0) {
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isMPEG = 1;
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} else if (strcmp(subsession->codecName(), "H263") == 0 ||
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strcmp(subsession->codecName(), "H263-1998") == 0) {
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bih->biCompression = sh_video->format
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= mmioFOURCC('H','2','6','3');
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} else if (strcmp(subsession->codecName(), "H261") == 0) {
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bih->biCompression = sh_video->format
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= mmioFOURCC('H','2','6','1');
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} else {
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fprintf(stderr,
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"Unknown mplayer format code for MIME type \"video/%s\"\n",
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subsession->codecName());
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}
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} else if (subsession == audioSubsession) {
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// Create a dummy audio stream header
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// to make the main mplayer code happy:
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sh_audio_t* sh_audio = new_sh_audio(demuxer,0);
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WAVEFORMATEX* wf = (WAVEFORMATEX*)calloc(1,sizeof(WAVEFORMATEX));
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sh_audio->wf = wf;
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demux_stream_t* d_audio = demuxer->audio;
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d_audio->sh = sh_audio; sh_audio->ds = d_audio;
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// Map known audio MIME types to the WAVEFORMATEX parameters
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// that this program uses. (Note that not all types need all
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// of the parameters to be set.)
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wf->nSamplesPerSec
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= subsession->rtpSource()->timestampFrequency(); // by default
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if (strcmp(subsession->codecName(), "MPA") == 0 ||
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strcmp(subsession->codecName(), "MPA-ROBUST") == 0 ||
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strcmp(subsession->codecName(), "X-MP3-DRAFT-00") == 0) {
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wf->wFormatTag = sh_audio->format = 0x55;
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// Note: 0x55 is for layer III, but should work for I,II also
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wf->nSamplesPerSec = 0; // sample rate is deduced from the data
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} else if (strcmp(subsession->codecName(), "AC3") == 0) {
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wf->wFormatTag = sh_audio->format = 0x2000;
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wf->nSamplesPerSec = 0; // sample rate is deduced from the data
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} else if (strcmp(subsession->codecName(), "PCMU") == 0) {
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wf->wFormatTag = sh_audio->format = 0x7;
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wf->nChannels = 1;
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wf->nAvgBytesPerSec = 8000;
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wf->nBlockAlign = 1;
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wf->wBitsPerSample = 8;
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wf->cbSize = 0;
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} else if (strcmp(subsession->codecName(), "PCMA") == 0) {
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wf->wFormatTag = sh_audio->format = 0x6;
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wf->nChannels = 1;
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wf->nAvgBytesPerSec = 8000;
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wf->nBlockAlign = 1;
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wf->wBitsPerSample = 8;
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wf->cbSize = 0;
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} else if (strcmp(subsession->codecName(), "GSM") == 0) {
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wf->wFormatTag = sh_audio->format = mmioFOURCC('a','g','s','m');
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wf->nChannels = 1;
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wf->nAvgBytesPerSec = 1650;
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wf->nBlockAlign = 33;
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wf->wBitsPerSample = 16;
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wf->cbSize = 0;
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} else if (strcmp(subsession->codecName(), "MP4A-LATM") == 0) {
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wf->wFormatTag = sh_audio->format = mmioFOURCC('m','p','4','a');
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#ifndef HAVE_FAAD
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fprintf(stderr, "WARNING: Playing MPEG-4 (AAC) Audio requires the \"faad\" library!\n");
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#endif
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#if (LIVEMEDIA_LIBRARY_VERSION_INT < 1042761600)
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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");
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#else
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// For the codec to work correctly, it needs "AudioSpecificConfig"
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// data, which is parsed from the "StreamMuxConfig" string that
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// was present (hopefully) in the SDP description:
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unsigned codecdata_len;
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sh_audio->codecdata
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= parseStreamMuxConfigStr(subsession->fmtp_config(),
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codecdata_len);
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sh_audio->codecdata_len = codecdata_len;
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#endif
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} else {
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fprintf(stderr,
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"Unknown mplayer format code for MIME type \"audio/%s\"\n",
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subsession->codecName());
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}
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// MPlayer codec-specific initialization on it:
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if (streamType == 0) { // video
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rtpState->videoBufferQueue
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= new ReadBufferQueue(subsession, demuxer, "video");
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rtpCodecInitialize_video(demuxer, subsession, flags);
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} else { // audio
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rtpState->audioBufferQueue
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= new ReadBufferQueue(subsession, demuxer, "audio");
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rtpCodecInitialize_audio(demuxer, subsession, flags);
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}
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}
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}
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// Hack: Create a 'RTPState' structure containing the state that
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// we just created, and store it in the demuxer's 'priv' field:
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RTPState* rtpState = new RTPState;
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rtpState->sdpDescription = sdpDescription;
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rtpState->rtspClient = rtspClient;
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rtpState->mediaSession = mediaSession;
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rtpState->audioBufferQueue
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= new ReadBufferQueue(audioSubsession, demuxer, "audio");
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rtpState->videoBufferQueue
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= new ReadBufferQueue(videoSubsession, demuxer, "video");
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rtpState->isMPEG = isMPEG;
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rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
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demuxer->priv = rtpState;
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rtpState->flags = flags;
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} while (0);
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}
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extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
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// Get the RTP state that was stored in the demuxer's 'priv' field:
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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return rtpState->isMPEG;
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return (rtpState->flags&RTPSTATE_IS_MPEG) != 0;
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}
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static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue,
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demux_stream_t* ds); // forward
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static ReadBuffer* getBuffer(ReadBufferQueue* bufferQueue,
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demuxer_t* demuxer); // forward
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extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
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// Get a filled-in "demux_packet" from the RTP source, and deliver it.
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@ -324,24 +228,46 @@ extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
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return 0;
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}
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// Check whether there's a full buffer to deliver to the client:
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bufferQueue->blockingFlag = 0;
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while (!deliverBufferIfAvailable(bufferQueue, ds)) {
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// Because we weren't able to deliver a buffer to the client immediately,
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// block myself until one comes available:
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TaskScheduler& scheduler
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= bufferQueue->readSource()->envir().taskScheduler();
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#if USAGEENVIRONMENT_LIBRARY_VERSION_INT >= 1038614400
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scheduler.doEventLoop(&bufferQueue->blockingFlag);
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#else
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scheduler.blockMyself(&bufferQueue->blockingFlag);
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#endif
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}
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ReadBuffer* readBuffer = getBuffer(bufferQueue, demuxer); // blocking
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if (readBuffer != NULL) ds_add_packet(ds, readBuffer->dp());
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if (demuxer->stream->eof) return 0; // source stream has closed down
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return 1;
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}
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Boolean awaitRTPPacket(demuxer_t* demuxer, unsigned streamType,
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unsigned char*& packetData, unsigned& packetDataLen) {
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// Begin by finding the buffer queue that we want to read from:
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// (Get this from the RTP state, which we stored in
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// the demuxer's 'priv' field)
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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ReadBufferQueue* bufferQueue = NULL;
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if (streamType == 0) {
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bufferQueue = rtpState->videoBufferQueue;
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} else if (streamType == 1) {
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bufferQueue = rtpState->audioBufferQueue;
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} else {
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fprintf(stderr, "awaitRTPPacket: internal error: unknown streamType %d\n",
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streamType);
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return False;
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}
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if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
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fprintf(stderr, "awaitRTPPacket failed: no appropriate RTP subsession has been set up\n");
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return False;
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}
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ReadBuffer* readBuffer = getBuffer(bufferQueue, demuxer); // blocking
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if (readBuffer == NULL) return False;
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demux_packet_t* dp = readBuffer->dp();
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packetData = dp->buffer;
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packetDataLen = dp->len;
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return True;
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}
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extern "C" void demux_close_rtp(demuxer_t* demuxer) {
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// Reclaim all RTP-related state:
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@ -366,24 +292,6 @@ extern "C" void demux_close_rtp(demuxer_t* demuxer) {
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////////// Extra routines that help implement the above interface functions:
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static void scheduleNewBufferRead(ReadBufferQueue* bufferQueue); // forward
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static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue,
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demux_stream_t* ds) {
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Boolean deliveredBuffer = False;
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ReadBuffer* readBuffer = bufferQueue->dequeue();
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if (readBuffer != NULL) {
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// Append the packet to the reader's DS stream:
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ds_add_packet(ds, readBuffer->dp());
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deliveredBuffer = True;
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}
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// Arrange to read a new packet into this queue:
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scheduleNewBufferRead(bufferQueue);
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return deliveredBuffer;
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}
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static void afterReading(void* clientData, unsigned frameSize,
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struct timeval presentationTime); // forward
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static void onSourceClosure(void* clientData); // forward
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@ -444,7 +352,7 @@ static void afterReading(void* clientData, unsigned frameSize,
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delete readBuffer;
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}
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// Signal any pending 'blockMyself()' call on this queue:
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// Signal any pending 'doEventLoop()' call on this queue:
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bufferQueue->blockingFlag = ~0;
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// Finally, arrange to do another read, if appropriate
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@ -458,10 +366,40 @@ static void onSourceClosure(void* clientData) {
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demuxer->stream->eof = 1;
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// Signal any pending 'blockMyself()' call on this queue:
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// Signal any pending 'doEventLoop()' call on this queue:
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bufferQueue->blockingFlag = ~0;
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}
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static ReadBuffer* getBufferIfAvailable(ReadBufferQueue* bufferQueue) {
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ReadBuffer* readBuffer = bufferQueue->dequeue();
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// Arrange to read a new packet into this queue:
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scheduleNewBufferRead(bufferQueue);
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return readBuffer;
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}
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static ReadBuffer* getBuffer(ReadBufferQueue* bufferQueue,
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demuxer_t* demuxer) {
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// Check whether there's a full buffer to deliver to the client:
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bufferQueue->blockingFlag = 0;
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ReadBuffer* readBuffer;
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while ((readBuffer = getBufferIfAvailable(bufferQueue)) == NULL
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&& !demuxer->stream->eof) {
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// Because we weren't able to deliver a buffer to the client immediately,
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// block myself until one comes available:
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TaskScheduler& scheduler
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= bufferQueue->readSource()->envir().taskScheduler();
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#if USAGEENVIRONMENT_LIBRARY_VERSION_INT >= 1038614400
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scheduler.doEventLoop(&bufferQueue->blockingFlag);
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#else
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scheduler.blockMyself(&bufferQueue->blockingFlag);
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#endif
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}
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return readBuffer;
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}
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||||
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////////// "ReadBuffer" and "ReadBufferQueue" implementation:
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#define MAX_QUEUE_SIZE 5
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|
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@ -0,0 +1,203 @@
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////////// Codec-specific routines used to interface between "MPlayer"
|
||||
////////// and the "LIVE.COM Streaming Media" libraries:
|
||||
|
||||
#include "demux_rtp_internal.h"
|
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extern "C" {
|
||||
#include "stheader.h"
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||||
}
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|
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static Boolean
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parseQTState_video(QuickTimeGenericRTPSource::QTState const& qtState,
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unsigned& fourcc); // forward
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static Boolean
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parseQTState_audio(QuickTimeGenericRTPSource::QTState const& qtState,
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unsigned& fourcc, unsigned& numChannels); // forward
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void rtpCodecInitialize_video(demuxer_t* demuxer,
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MediaSubsession* subsession,
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unsigned& flags) {
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flags = 0;
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// Create a dummy video stream header
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||||
// to make the main MPlayer code happy:
|
||||
sh_video_t* sh_video = new_sh_video(demuxer,0);
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||||
BITMAPINFOHEADER* bih
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= (BITMAPINFOHEADER*)calloc(1,sizeof(BITMAPINFOHEADER));
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bih->biSize = sizeof(BITMAPINFOHEADER);
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||||
sh_video->bih = bih;
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demux_stream_t* d_video = demuxer->video;
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d_video->sh = sh_video; sh_video->ds = d_video;
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||||
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||||
// If we happen to know the subsession's video frame rate, set it,
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||||
// 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;
|
||||
}
|
|
@ -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
|
Loading…
Reference in New Issue