RTC: Remove dead code

pull/1804/head
winlin 5 years ago
parent e446cf466f
commit 431f852e05

2
trunk/configure vendored

@ -279,7 +279,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
"srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
"srs_app_coworkers" "srs_app_hybrid")
if [[ $SRS_RTC == YES ]]; then
MODULE_FILES+=("srs_app_rtc" "srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_codec" "srs_app_rtc_sdp"
MODULE_FILES+=("srs_app_rtc_conn" "srs_app_rtc_dtls" "srs_app_rtc_codec" "srs_app_rtc_sdp"
"srs_app_rtc_queue" "srs_app_rtc_server" "srs_app_rtc_source")
fi
if [[ $SRS_GB28181 == YES ]]; then

@ -1,204 +0,0 @@
/**
* The MIT License (MIT)
*
* Copyright (c) 2013-2020 John
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
* the Software, and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include <srs_app_rtc.hpp>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <unistd.h>
#include <algorithm>
#include <sstream>
using namespace std;
#include <srs_kernel_buffer.hpp>
#include <srs_kernel_error.hpp>
#include <srs_kernel_codec.hpp>
#include <srs_kernel_flv.hpp>
#include <srs_kernel_rtc_rtp.hpp>
#include <srs_app_config.hpp>
#include <srs_app_source.hpp>
#include <srs_core_autofree.hpp>
#include <srs_app_pithy_print.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_kernel_codec.hpp>
#include <srs_kernel_file.hpp>
#include <srs_app_utility.hpp>
#include <srs_app_http_hooks.hpp>
#include <srs_protocol_format.hpp>
#include <srs_rtmp_stack.hpp>
#include <openssl/rand.h>
#include <srs_app_rtc_codec.hpp>
// TODO: Add this function into SrsRtpMux class.
extern srs_error_t aac_raw_append_adts_header(SrsSharedPtrMessage* shared_audio, SrsFormat* format, char** pbuf, int* pnn_buf);
SrsRtpH264Muxer::SrsRtpH264Muxer()
{
discard_bframe = false;
}
SrsRtpH264Muxer::~SrsRtpH264Muxer()
{
}
srs_error_t SrsRtpH264Muxer::filter(SrsSharedPtrMessage* shared_frame, SrsFormat* format)
{
srs_error_t err = srs_success;
// If IDR, we will insert SPS/PPS before IDR frame.
if (format->video && format->video->has_idr) {
shared_frame->set_has_idr(true);
}
// Update samples to shared frame.
for (int i = 0; i < format->video->nb_samples; ++i) {
SrsSample* sample = &format->video->samples[i];
// Because RTC does not support B-frame, so we will drop them.
// TODO: Drop B-frame in better way, which not cause picture corruption.
if (discard_bframe) {
if ((err = sample->parse_bframe()) != srs_success) {
return srs_error_wrap(err, "parse bframe");
}
if (sample->bframe) {
continue;
}
}
}
if (format->video->nb_samples <= 0) {
return err;
}
shared_frame->set_samples(format->video->samples, format->video->nb_samples);
return err;
}
SrsRtc::SrsRtc()
{
req = NULL;
enabled = false;
disposable = false;
last_update_time = 0;
discard_aac = false;
}
SrsRtc::~SrsRtc()
{
srs_freep(rtp_h264_muxer);
}
void SrsRtc::dispose()
{
if (enabled) {
on_unpublish();
}
}
// TODO: FIXME: Dead code?
srs_error_t SrsRtc::cycle()
{
srs_error_t err = srs_success;
return err;
}
srs_error_t SrsRtc::initialize(SrsRequest* r)
{
srs_error_t err = srs_success;
req = r;
rtp_h264_muxer = new SrsRtpH264Muxer();
rtp_h264_muxer->discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
// TODO: FIXME: Support reload and log it.
discard_aac = _srs_config->get_rtc_aac_discard(req->vhost);
return err;
}
srs_error_t SrsRtc::on_publish()
{
srs_error_t err = srs_success;
// update the hls time, for hls_dispose.
last_update_time = srs_get_system_time();
// support multiple publish.
if (enabled) {
return err;
}
if (!_srs_config->get_rtc_enabled(req->vhost)) {
return err;
}
// if enabled, open the muxer.
enabled = true;
// ok, the hls can be dispose, or need to be dispose.
disposable = true;
return err;
}
void SrsRtc::on_unpublish()
{
// support multiple unpublish.
if (!enabled) {
return;
}
enabled = false;
}
srs_error_t SrsRtc::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
{
srs_error_t err = srs_success;
// TODO: FIXME: Maybe it should config on vhost level.
if (!enabled) {
return err;
}
// Ignore if no format->vcodec, it means the codec is not parsed, or unknown codec.
// @issue https://github.com/ossrs/srs/issues/1506#issuecomment-562079474
if (!format->vcodec) {
return err;
}
// update the hls time, for hls_dispose.
last_update_time = srs_get_system_time();
// ignore info frame,
// @see https://github.com/ossrs/srs/issues/288#issuecomment-69863909
srs_assert(format->video);
return rtp_h264_muxer->filter(shared_video, format);
}

@ -1,86 +0,0 @@
/**
* The MIT License (MIT)
*
* Copyright (c) 2013-2020 John
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
* the Software, and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#ifndef SRS_APP_RTC_HPP
#define SRS_APP_RTC_HPP
#include <srs_core.hpp>
#include <string>
#include <vector>
#include <map>
class SrsFormat;
class SrsSample;
class SrsSharedPtrMessage;
class SrsRequest;
class SrsOriginHub;
class SrsAudioRecode;
class SrsBuffer;
// The RTP packet max size, should never exceed this size.
const int kRtpPacketSize = 1500;
// Payload type will rewrite in srs_app_rtc_conn.cpp when send to client.
const uint8_t kOpusPayloadType = 111;
const uint8_t kH264PayloadType = 102;
// SSRC will rewrite in srs_app_rtc_conn.cpp when send to client.
const uint32_t kAudioSSRC = 1;
const uint32_t kVideoSSRC = 2;
// TODO: Define interface class like ISrsRtpMuxer
class SrsRtpH264Muxer
{
public:
bool discard_bframe;
public:
SrsRtpH264Muxer();
virtual ~SrsRtpH264Muxer();
public:
srs_error_t filter(SrsSharedPtrMessage* shared_video, SrsFormat* format);
};
class SrsRtc
{
private:
SrsRequest* req;
bool enabled;
bool disposable;
bool discard_aac;
srs_utime_t last_update_time;
SrsRtpH264Muxer* rtp_h264_muxer;
public:
SrsRtc();
virtual ~SrsRtc();
public:
virtual void dispose();
virtual srs_error_t cycle();
public:
virtual srs_error_t initialize(SrsRequest* r);
virtual srs_error_t on_publish();
virtual void on_unpublish();
virtual srs_error_t on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format);
};
#endif

@ -473,7 +473,6 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
source_ = source;
meta = new SrsMetaCache();
format = new SrsRtmpFormat();
rtc = new SrsRtc();
codec = new SrsAudioRecode(kChannel, kSamplerate);
discard_aac = false;
discard_bframe = false;
@ -483,7 +482,6 @@ SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
{
srs_freep(meta);
srs_freep(format);
srs_freep(rtc);
srs_freep(codec);
}
@ -497,10 +495,6 @@ srs_error_t SrsRtcFromRtmpBridger::initialize(SrsRequest* r)
return srs_error_wrap(err, "format initialize");
}
if ((err = rtc->initialize(r)) != srs_success) {
return srs_error_wrap(err, "rtc initialize");
}
if ((err = codec->initialize()) != srs_success) {
return srs_error_wrap(err, "init codec");
}
@ -522,10 +516,6 @@ srs_error_t SrsRtcFromRtmpBridger::on_publish()
{
srs_error_t err = srs_success;
if ((err = rtc->on_publish()) != srs_success) {
return srs_error_wrap(err, "rtc publish");
}
// TODO: FIXME: Should sync with bridger?
if ((err = source_->on_publish()) != srs_success) {
return srs_error_wrap(err, "source publish");
@ -543,8 +533,6 @@ void SrsRtcFromRtmpBridger::on_unpublish()
// TODO: FIXME: Should sync with bridger?
source_->on_unpublish();
rtc->on_unpublish();
// Reset the metadata cache, to make VLC happy when disable/enable stream.
// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
meta->update_previous_vsh();
@ -667,14 +655,6 @@ srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
return srs_error_wrap(err, "format consume video");
}
// Parse RTMP message to RTP packets, in FU-A if too large.
if ((err = rtc->on_video(msg, format)) != srs_success) {
// TODO: We should support more strategies.
srs_warn("rtc: ignore video error %s", srs_error_desc(err).c_str());
srs_error_reset(err);
rtc->on_unpublish();
}
return source_->on_video_imp(msg);
}

@ -180,8 +180,6 @@ private:
bool discard_aac;
SrsAudioRecode* codec;
bool discard_bframe;
// rtc handler
SrsRtc* rtc;
public:
SrsRtcFromRtmpBridger(SrsRtcSource* source);
virtual ~SrsRtcFromRtmpBridger();

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