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https://github.com/ZLMediaKit/ZLMediaKit.git
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Merge branch 'fix_rtc_g711_play' of github.com:ZLMediaKit/ZLMediaKit
This commit is contained in:
commit
96061f1d36
@ -18,6 +18,7 @@
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#include "AACRtp.h"
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#include "H265Rtp.h"
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#include "CommonRtp.h"
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#include "G711Rtp.h"
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#include "Opus.h"
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#include "G711.h"
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#include "L16.h"
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@ -119,9 +120,13 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
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case CodecH265 : return std::make_shared<H265RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
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case CodecAAC : return std::make_shared<AACRtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
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case CodecL16 :
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case CodecOpus :
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case CodecOpus : return std::make_shared<CommonRtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
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case CodecG711A :
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case CodecG711U : return std::make_shared<CommonRtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
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case CodecG711U :
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if(pt == Rtsp::PT_PCMA || pt == Rtsp::PT_PCMU){
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return std::make_shared<G711RtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved,1);
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}
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return std::make_shared<CommonRtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
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default : WarnL << "暂不支持该CodecId:" << codec_id; return nullptr;
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}
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}
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46
src/Extension/G711Rtp.cpp
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46
src/Extension/G711Rtp.cpp
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@ -0,0 +1,46 @@
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#include "G711Rtp.h"
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G711RtpEncoder::G711RtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_size,
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uint32_t sample_rate, uint8_t payload_type, uint8_t interleaved,uint32_t channels)
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: CommonRtpDecoder(codec), RtpInfo(ssrc, mtu_size, sample_rate, payload_type, interleaved) {
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_cache_frame = FrameImp::create();
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_cache_frame->_codec_id = codec;
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_channels = channels;
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}
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bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame){
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auto dur = (_cache_frame->size()-_cache_frame->prefixSize())/(8*_channels);
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auto next_pts = _cache_frame->pts()+dur;
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if(next_pts == 0){
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_cache_frame->_pts = frame->pts();
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}else{
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if((next_pts+20) < frame->pts()){// 有丢包超过20ms
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_cache_frame->_pts = frame->pts() - dur;
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}
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}
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_cache_frame->_buffer.append(frame->data() + frame->prefixSize(),frame->size() - frame->prefixSize());
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auto stamp = _cache_frame->pts();
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auto ptr = _cache_frame->data() + _cache_frame->prefixSize();
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auto len = _cache_frame->size() - _cache_frame->prefixSize();
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auto remain_size = len;
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auto max_size = 160*_channels; //20 ms per rtp
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int n = 0;
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bool mark = false;
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while (remain_size >= max_size) {
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size_t rtp_size;
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if (remain_size >= max_size) {
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rtp_size = max_size;
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} else {
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break;
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}
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n++;
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stamp += 20;
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RtpCodec::inputRtp(makeRtp(getTrackType(), ptr, rtp_size, mark, stamp), false);
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ptr += rtp_size;
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remain_size -= rtp_size;
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}
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_cache_frame->_buffer.erase(0,n*max_size);
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_cache_frame->_pts += 20*n;
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return len > 0;
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}
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51
src/Extension/G711Rtp.h
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51
src/Extension/G711Rtp.h
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@ -0,0 +1,51 @@
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/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef ZLMEDIAKIT_G711RTP_H
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#define ZLMEDIAKIT_G711RTP_H
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#include "Frame.h"
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#include "CommonRtp.h"
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#include "Rtsp/RtpCodec.h"
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namespace mediakit{
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/**
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* G711 rtp编码类
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*/
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class G711RtpEncoder : public CommonRtpDecoder, public RtpInfo {
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public:
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typedef std::shared_ptr <G711RtpEncoder> Ptr;
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~G711RtpEncoder() override {}
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/**
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* 构造函数
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* @param codec 编码类型
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* @param ssrc ssrc
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* @param mtu_size mtu 大小
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* @param sample_rate 采样率
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* @param payload_type pt类型
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* @param interleaved rtsp interleaved 值
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*/
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G711RtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_size, uint32_t sample_rate, uint8_t payload_type, uint8_t interleaved,uint32_t channels);
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/**
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* 输入帧数据并编码成rtp
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*/
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bool inputFrame(const Frame::Ptr &frame) override;
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private:
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FrameImp::Ptr _cache_frame;
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uint32_t _channels = 1;
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};
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}//namespace mediakit
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#endif //ZLMEDIAKIT_G711RTP_H
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@ -43,6 +43,7 @@ void WebRtcPlayer::onStartWebRTC() {
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}
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size_t i = 0;
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pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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//TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
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strongSelf->onSendRtp(rtp, ++i == pkt->size());
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});
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});
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@ -57,7 +58,6 @@ void WebRtcPlayer::onStartWebRTC() {
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//使用完毕后,释放强引用,这样确保推流器断开后能及时注销媒体
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_play_src = nullptr;
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}
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void WebRtcPlayer::onDestory() {
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WebRtcTransportImp::onDestory();
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