/*
The MIT License (MIT)

Copyright (c) 2013-2014 winlin

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_hls.hpp>

/**
* the public data, event HLS disable, others can use it.
*/
// 0 = 5.5 kHz = 5512 Hz
// 1 = 11 kHz = 11025 Hz
// 2 = 22 kHz = 22050 Hz
// 3 = 44 kHz = 44100 Hz
int flv_sample_rates[] = {5512, 11025, 22050, 44100};

// the sample rates in the codec,
// in the sequence header.
int aac_sample_rates[] = 
{
    96000, 88200, 64000, 48000,
    44100, 32000, 24000, 22050,
    16000, 12000, 11025,  8000,
    7350,     0,     0,    0
};

/**
* the HLS section, only available when HLS enabled.
*/
#ifdef SRS_AUTO_HLS

#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <string.h>

#include <algorithm>
using namespace std;

#include <srs_kernel_error.hpp>
#include <srs_kernel_codec.hpp>
#include <srs_protocol_amf0.hpp>
#include <srs_protocol_stack.hpp>
#include <srs_app_config.hpp>
#include <srs_app_source.hpp>
#include <srs_core_autofree.hpp>
#include <srs_protocol_rtmp.hpp>
#include <srs_app_pithy_print.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_app_avc_aac.hpp>
#include <srs_kernel_file.hpp>
#include <srs_kernel_buffer.hpp>

// max PES packets size to flush the video.
#define SRS_AUTO_HLS_AUDIO_CACHE_SIZE 1024 * 1024

// drop the segment when duration of ts too small.
#define SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS 100

// @see: NGX_RTMP_HLS_DELAY, 
// 63000: 700ms, ts_tbn=90000
// 72000: 800ms, ts_tbn=90000
// @see https://github.com/simple-rtmp-server/srs/issues/151#issuecomment-71352511
#define SRS_AUTO_HLS_DELAY 72000

// the mpegts header specifed the video/audio pid.
#define TS_VIDEO_PID 256
#define TS_AUDIO_PID 257

// ts aac stream id.
#define TS_AUDIO_AAC 0xc0
// ts avc stream id.
#define TS_VIDEO_AVC 0xe0

// @see: ngx_rtmp_hls_audio
/* We assume here AAC frame size is 1024
 * Need to handle AAC frames with frame size of 960 */
#define _SRS_AAC_SAMPLE_SIZE 1024

// in ms, for HLS aac sync time.
#define SRS_CONF_DEFAULT_AAC_SYNC 100
// in ms, for HLS aac flush the audio
#define SRS_CONF_DEFAULT_AAC_DELAY 100

// @see: ngx_rtmp_mpegts_header
u_int8_t mpegts_header[] = {
    /* TS */
    0x47, 0x40, 0x00, 0x10, 0x00,
    /* PSI */
    0x00, 0xb0, 0x0d, 0x00, 0x01, 0xc1, 0x00, 0x00,
    /* PAT */
    0x00, 0x01, 0xf0, 0x01,
    /* CRC */
    0x2e, 0x70, 0x19, 0x05,
    /* stuffing 167 bytes */
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    
    /* TS */
    0x47, 0x50, 0x01, 0x10, 0x00,
    /* PSI */
    0x02, 0xb0, 0x17, 0x00, 0x01, 0xc1, 0x00, 0x00,
    /* PMT */
    0xe1, 0x00,
    0xf0, 0x00,
    // must generate header with/without video, @see:
    // https://github.com/simple-rtmp-server/srs/issues/40
    0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
    0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
    /*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
    /* CRC */
    0x2f, 0x44, 0xb9, 0x9b, /* crc for aac */
    /*0x4e, 0x59, 0x3d, 0x1e,*/ /* crc for mp3 */
    /* stuffing 157 bytes */
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
};

// @see: ngx_rtmp_SrsMpegtsFrame_t
class SrsMpegtsFrame
{
public:
    int64_t         pts;
    int64_t         dts;
    int             pid;
    int             sid;
    int             cc;
    bool            key;
    
    SrsMpegtsFrame()
    {
        pts = dts = 0;
        pid = sid = cc = 0;
        key = false;
    }
};

// @see: ngx_rtmp_mpegts.c
// TODO: support full mpegts feature in future.
class SrsMpegtsWriter
{
public:
    static int write_header(SrsFileWriter* writer)
    {
        int ret = ERROR_SUCCESS;
        
        if ((ret = writer->write(mpegts_header, sizeof(mpegts_header), NULL)) != ERROR_SUCCESS) {
            ret = ERROR_HLS_WRITE_FAILED;
            srs_error("write ts file header failed. ret=%d", ret);
            return ret;
        }

        return ret;
    }
    static int write_frame(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsBuffer* buffer)
    {
        int ret = ERROR_SUCCESS;
        
        if (!buffer->bytes() || buffer->length() <= 0) {
            return ret;
        }
        
        char* last = buffer->bytes() + buffer->length();
        char* pos = buffer->bytes();
        
        bool first = true;
        while (pos < last) {
            static char packet[188];
            char* p = packet;
            
            frame->cc++;
            
            // sync_byte; //8bits
            *p++ = 0x47;
            // pid; //13bits
            *p++ = (frame->pid >> 8) & 0x1f;
            // payload_unit_start_indicator; //1bit
            if (first) {
                p[-1] |= 0x40;
            }
            *p++ = frame->pid;
            
            // transport_scrambling_control; //2bits
            // adaption_field_control; //2bits, 0x01: PayloadOnly
            // continuity_counter; //4bits
            *p++ = 0x10 | (frame->cc & 0x0f);
            
            if (first) {
                first = false;
                if (frame->key) {
                    p[-1] |= 0x20; // Both Adaption and Payload
                    *p++ = 7;    // size
                    *p++ = 0x50; // random access + PCR
                    // about the pcr, read https://github.com/simple-rtmp-server/srs/issues/151#issuecomment-71352511
                    p = write_pcr(p, frame->dts);
                }
                
                // PES header
                // packet_start_code_prefix; //24bits, '00 00 01'
                *p++ = 0x00;
                *p++ = 0x00;
                *p++ = 0x01;
                //8bits
                *p++ = frame->sid;
                
                // pts(33bits) need 5bytes.
                u_int8_t header_size = 5;
                u_int8_t flags = 0x80; // pts
                
                // dts(33bits) need 5bytes also
                if (frame->dts != frame->pts) {
                    header_size += 5;
                    flags |= 0x40; // dts
                }
                
                // 3bytes: flag fields from PES_packet_length to PES_header_data_length
                int pes_size = (last - pos) + header_size + 3;
                if (pes_size > 0xffff) {
                    /**
                    * when actual packet length > 0xffff(65535),
                    * which exceed the max u_int16_t packet length,
                    * use 0 packet length, the next unit start indicates the end of packet.
                    */
                    pes_size = 0;
                }
                
                // PES_packet_length; //16bits
                *p++ = (pes_size >> 8);
                *p++ = pes_size;
                
                // PES_scrambling_control; //2bits, '10'
                // PES_priority; //1bit
                // data_alignment_indicator; //1bit
                // copyright; //1bit
                // original_or_copy; //1bit    
                *p++ = 0x80; /* H222 */
                
                // PTS_DTS_flags; //2bits
                // ESCR_flag; //1bit
                // ES_rate_flag; //1bit
                // DSM_trick_mode_flag; //1bit
                // additional_copy_info_flag; //1bit
                // PES_CRC_flag; //1bit
                // PES_extension_flag; //1bit
                *p++ = flags;
                
                // PES_header_data_length; //8bits
                *p++ = header_size;

                // pts; // 33bits
                p = write_pts(p, flags >> 6, frame->pts + SRS_AUTO_HLS_DELAY);
                
                // dts; // 33bits
                if (frame->dts != frame->pts) {
                    p = write_pts(p, 1, frame->dts + SRS_AUTO_HLS_DELAY);
                }
            }
            
            int body_size = sizeof(packet) - (p - packet);
            int in_size = last - pos;
            
            if (body_size <= in_size) {
                memcpy(p, pos, body_size);
                pos += body_size;
            } else {
                p = fill_stuff(p, packet, body_size, in_size);
                memcpy(p, pos, in_size);
                pos = last;
            }
            
            // write ts packet
            if ((ret = writer->write(packet, sizeof(packet), NULL)) != ERROR_SUCCESS) {
                ret = ERROR_HLS_WRITE_FAILED;
                srs_error("write ts file failed. ret=%d", ret);
                return ret;
            }
        }
        
        return ret;
    }
private:
    static char* fill_stuff(char* pes_body_end, char* packet, int body_size, int in_size)
    {
        char* p = pes_body_end;
        
        // insert the stuff bytes before PES body
        int stuff_size = (body_size - in_size);
        
        // adaption_field_control; //2bits
        if (packet[3] & 0x20) {
            //  has adaptation
            // packet[4]: adaption_field_length
            // packet[5]: adaption field data
            // base: start of PES body
            char* base = &packet[5] + packet[4];
            int len = p - base;
            p = (char*)memmove(base + stuff_size, base, len) + len;
            // increase the adaption field size.
            packet[4] += stuff_size;
            
            return p;
        }

        // create adaption field.
        // adaption_field_control; //2bits
        packet[3] |= 0x20;
        // base: start of PES body
        char* base = &packet[4];
        int len = p - base;
        p = (char*)memmove(base + stuff_size, base, len) + len;
        // adaption_field_length; //8bits
        packet[4] = (stuff_size - 1);
        if (stuff_size >= 2) {
            // adaption field flags.
            packet[5] = 0;
            // adaption data.
            if (stuff_size > 2) {
                memset(&packet[6], 0xff, stuff_size - 2);
            }
        }
        
        return p;
    }
    static char* write_pcr(char* p, int64_t pcr)
    {
        // the pcr=dts-delay, where dts = frame->dts + delay
        // and the pcr should never be negative
        // @see https://github.com/simple-rtmp-server/srs/issues/268
        srs_assert(pcr >= 0);
        
        int64_t v = pcr;
        
        *p++ = (char) (v >> 25);
        *p++ = (char) (v >> 17);
        *p++ = (char) (v >> 9);
        *p++ = (char) (v >> 1);
        *p++ = (char) (v << 7 | 0x7e);
        *p++ = 0;
    
        return p;
    }
    static char* write_pts(char* p, u_int8_t fb, int64_t pts)
    {
        int32_t val;
    
        val = fb << 4 | (((pts >> 30) & 0x07) << 1) | 1;
        *p++ = val;
    
        val = (((pts >> 15) & 0x7fff) << 1) | 1;
        *p++ = (val >> 8);
        *p++ = val;
    
        val = (((pts) & 0x7fff) << 1) | 1;
        *p++ = (val >> 8);
        *p++ = val;
    
        return p;
    }
};

SrsHlsAacJitter::~SrsHlsAacJitter()
{
}

int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate)
{
    // use sample rate in flv/RTMP.
    int flv_sample_rate = flv_sample_rates[sample_rate & 0x03];

    // override the sample rate by sequence header
    if (aac_sample_rate != __SRS_AAC_SAMPLE_RATE_UNSET) {
        flv_sample_rate = aac_sample_rates[aac_sample_rate];
    }

    // sync time set to 0, donot adjust the aac timestamp.
    if (!sync_ms) {
        return flv_pts;
    }
    
    // @see: ngx_rtmp_hls_audio
    // drop the rtmp audio packet timestamp, re-calc it by sample rate.
    // 
    // resample for the tbn of ts is 90000, flv is 1000,
    // we will lost timestamp if use audio packet timestamp,
    // so we must resample. or audio will corupt in IOS.
    int64_t est_pts = base_pts + nb_samples * 90000LL * _SRS_AAC_SAMPLE_SIZE / flv_sample_rate;
    int64_t dpts = (int64_t) (est_pts - flv_pts);

    if (dpts <= (int64_t) sync_ms * 90 && dpts >= (int64_t) sync_ms * -90) {
        srs_info("HLS correct aac pts "
            "from %"PRId64" to %"PRId64", base=%"PRId64", nb_samples=%d, sample_rate=%d",
            flv_pts, est_pts, nb_samples, flv_sample_rate, base_pts);

        nb_samples++;
        
        return est_pts;
    }
    
    // resync
    srs_trace("HLS aac resync, dpts=%"PRId64", pts=%"PRId64
        ", base=%"PRId64", nb_samples=%"PRId64", sample_rate=%d",
        dpts, flv_pts, base_pts, nb_samples, flv_sample_rate);
    
    base_pts = flv_pts;
    nb_samples = 1;
    
    return flv_pts;
}

void SrsHlsAacJitter::on_buffer_continue()
{
    nb_samples++;
}

SrsTSMuxer::SrsTSMuxer()
{
    writer = new SrsFileWriter();
}

SrsTSMuxer::~SrsTSMuxer()
{
    close();
    srs_freep(writer);
}

int SrsTSMuxer::open(string _path)
{
    int ret = ERROR_SUCCESS;
    
    path = _path;
    
    close();
    
    if ((ret = writer->open(path)) != ERROR_SUCCESS) {
        return ret;
    }

    // write mpegts header
    if ((ret = SrsMpegtsWriter::write_header(writer)) != ERROR_SUCCESS) {
        return ret;
    }
    
    return ret;
}

int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsBuffer* ab)
{
    int ret = ERROR_SUCCESS;
    
    if ((ret = SrsMpegtsWriter::write_frame(writer, af, ab)) != ERROR_SUCCESS) {
        return ret;
    }
    
    return ret;
}

int SrsTSMuxer::write_video(SrsMpegtsFrame* vf, SrsBuffer* vb)
{
    int ret = ERROR_SUCCESS;
    
    if ((ret = SrsMpegtsWriter::write_frame(writer, vf, vb)) != ERROR_SUCCESS) {
        return ret;
    }
    
    return ret;
}

void SrsTSMuxer::close()
{
    writer->close();
}

SrsHlsSegment::SrsHlsSegment()
{
    duration = 0;
    sequence_no = 0;
    muxer = new SrsTSMuxer();
    segment_start_dts = 0;
    is_sequence_header = false;
}

SrsHlsSegment::~SrsHlsSegment()
{
    srs_freep(muxer);
}

void SrsHlsSegment::update_duration(int64_t current_frame_dts)
{
    // we use video/audio to update segment duration,
    // so when reap segment, some previous audio frame will
    // update the segment duration, which is nagetive,
    // just ignore it.
    if (current_frame_dts < segment_start_dts) {
        return;
    }
    
    duration = (current_frame_dts - segment_start_dts) / 90000.0;
    srs_assert(duration >= 0);
    
    return;
}

SrsHlsAacJitter::SrsHlsAacJitter()
{
    base_pts = 0;
    nb_samples = 0;

    // TODO: config it, 0 means no adjust
    sync_ms = SRS_CONF_DEFAULT_AAC_SYNC;
}

SrsHlsMuxer::SrsHlsMuxer()
{
    hls_fragment = hls_window = 0;
    _sequence_no = 0;
    current = NULL;
}

SrsHlsMuxer::~SrsHlsMuxer()
{
    std::vector<SrsHlsSegment*>::iterator it;
    for (it = segments.begin(); it != segments.end(); ++it) {
        SrsHlsSegment* segment = *it;
        srs_freep(segment);
    }
    segments.clear();
    
    srs_freep(current);
}

int SrsHlsMuxer::sequence_no()
{
    return _sequence_no;
}

int SrsHlsMuxer::update_config(
    string _app, string _stream, string path, int fragment, int window
) {
    int ret = ERROR_SUCCESS;
    
    app = _app;
    stream = _stream;
    hls_path = path;
    hls_fragment = fragment;
    hls_window = window;
    
    return ret;
}

int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
{
    int ret = ERROR_SUCCESS;
    
    if (current) {
        srs_warn("ignore the segment open, for segment is already open.");
        return ret;
    }
    
    // TODO: create all parents dirs.
    // create dir for app.
    if ((ret = create_dir()) != ERROR_SUCCESS) {
        return ret;
    }
    
    // when segment open, the current segment must be NULL.
    srs_assert(!current);
    
    // new segment.
    current = new SrsHlsSegment();
    current->sequence_no = _sequence_no++;
    current->segment_start_dts = segment_start_dts;
    
    // generate filename.
    char filename[128];
    snprintf(filename, sizeof(filename), 
        "%s-%d.ts", stream.c_str(), current->sequence_no);
    
    // TODO: use temp file and rename it.
    current->full_path = hls_path;
    current->full_path += "/";
    current->full_path += app;
    current->full_path += "/";
    current->full_path += filename;
    
    // TODO: support base url, and so on.
    current->uri = filename;
    
    std::string tmp_file = current->full_path + ".tmp";
    if ((ret = current->muxer->open(tmp_file.c_str())) != ERROR_SUCCESS) {
        srs_error("open hls muxer failed. ret=%d", ret);
        return ret;
    }
    srs_info("open HLS muxer success. path=%s, tmp=%s", 
        current->full_path.c_str(), tmp_file.c_str());
    
    return ret;
}

int SrsHlsMuxer::on_sequence_header()
{
    int ret = ERROR_SUCCESS;
    
    srs_assert(current);
    
    // set the current segment to sequence header,
    // when close the segement, it will write a discontinuity to m3u8 file.
    current->is_sequence_header = true;
    
    return ret;
}

bool SrsHlsMuxer::is_segment_overflow()
{
    srs_assert(current);
    return current->duration >= hls_fragment;
}

bool SrsHlsMuxer::is_segment_absolutely_overflow()
{
    srs_assert(current);
    return current->duration >= 2 * hls_fragment;
}

int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsBuffer* ab)
{
    int ret = ERROR_SUCCESS;

    // if current is NULL, segment is not open, ignore the flush event.
    if (!current) {
        srs_warn("flush audio ignored, for segment is not open.");
        return ret;
    }
    
    if (ab->length() <= 0) {
        return ret;
    }
    
    // update the duration of segment.
    current->update_duration(af->pts);
    
    if ((ret = current->muxer->write_audio(af, ab)) != ERROR_SUCCESS) {
        return ret;
    }
    
    // write success, clear and free the buffer
    ab->erase(ab->length());

    return ret;
}

int SrsHlsMuxer::flush_video(SrsMpegtsFrame* /*af*/, SrsBuffer* /*ab*/, SrsMpegtsFrame* vf, SrsBuffer* vb)
{
    int ret = ERROR_SUCCESS;

    // if current is NULL, segment is not open, ignore the flush event.
    if (!current) {
        srs_warn("flush video ignored, for segment is not open.");
        return ret;
    }
    
    srs_assert(current);
    
    // update the duration of segment.
    current->update_duration(vf->dts);
    
    if ((ret = current->muxer->write_video(vf, vb)) != ERROR_SUCCESS) {
        return ret;
    }
    
    // write success, clear and free the buffer
    vb->erase(vb->length());
    
    return ret;
}

int SrsHlsMuxer::segment_close(string log_desc)
{
    int ret = ERROR_SUCCESS;
    
    if (!current) {
        srs_warn("ignore the segment close, for segment is not open.");
        return ret;
    }
    
    // when close current segment, the current segment must not be NULL.
    srs_assert(current);

    // assert segment duplicate.
    std::vector<SrsHlsSegment*>::iterator it;
    it = std::find(segments.begin(), segments.end(), current);
    srs_assert(it == segments.end());

    // valid, add to segments if segment duration is ok
    if (current->duration * 1000 >= SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
        segments.push_back(current);
    
        srs_info("%s reap ts segment, sequence_no=%d, uri=%s, duration=%.2f, start=%"PRId64"",
            log_desc.c_str(), current->sequence_no, current->uri.c_str(), current->duration, 
            current->segment_start_dts);
    
        // close the muxer of finished segment.
        srs_freep(current->muxer);
        std::string full_path = current->full_path;
        current = NULL;
        
        // rename from tmp to real path
        std::string tmp_file = full_path + ".tmp";
        if (rename(tmp_file.c_str(), full_path.c_str()) < 0) {
            ret = ERROR_HLS_WRITE_FAILED;
            srs_error("rename ts file failed, %s => %s. ret=%d", 
                tmp_file.c_str(), full_path.c_str(), ret);
            return ret;
        }
    } else {
        // reuse current segment index.
        _sequence_no--;
        
        srs_trace("%s drop ts segment, sequence_no=%d, uri=%s, duration=%.2f, start=%"PRId64"",
            log_desc.c_str(), current->sequence_no, current->uri.c_str(), current->duration, 
            current->segment_start_dts);
        
        // rename from tmp to real path
        std::string tmp_file = current->full_path + ".tmp";
        unlink(tmp_file.c_str());
        
        srs_freep(current);
    }
    
    // the segments to remove
    std::vector<SrsHlsSegment*> segment_to_remove;
    
    // shrink the segments.
    double duration = 0;
    int remove_index = -1;
    for (int i = segments.size() - 1; i >= 0; i--) {
        SrsHlsSegment* segment = segments[i];
        duration += segment->duration;
        
        if ((int)duration > hls_window) {
            remove_index = i;
            break;
        }
    }
    for (int i = 0; i < remove_index && !segments.empty(); i++) {
        SrsHlsSegment* segment = *segments.begin();
        segments.erase(segments.begin());
        segment_to_remove.push_back(segment);
    }
    
    // refresh the m3u8, donot contains the removed ts
    ret = refresh_m3u8();

    // remove the ts file.
    for (int i = 0; i < (int)segment_to_remove.size(); i++) {
        SrsHlsSegment* segment = segment_to_remove[i];
        unlink(segment->full_path.c_str());
        srs_freep(segment);
    }
    segment_to_remove.clear();
    
    // check ret of refresh m3u8
    if (ret != ERROR_SUCCESS) {
        srs_error("refresh m3u8 failed. ret=%d", ret);
        return ret;
    }
    
    return ret;
}

int SrsHlsMuxer::refresh_m3u8()
{
    int ret = ERROR_SUCCESS;
    
    std::string m3u8_file = hls_path;
    m3u8_file += "/";
    m3u8_file += app;
    m3u8_file += "/";
    m3u8_file += stream;
    m3u8_file += ".m3u8";
    
    m3u8 = m3u8_file;
    m3u8_file += ".temp";
    
    int fd = -1;
    ret = _refresh_m3u8(fd, m3u8_file);
    if (fd >= 0) {
        close(fd);
        if (rename(m3u8_file.c_str(), m3u8.c_str()) < 0) {
            ret = ERROR_HLS_WRITE_FAILED;
            srs_error("rename m3u8 file failed. "
                "%s => %s, ret=%d", m3u8_file.c_str(), m3u8.c_str(), ret);
        }
    }
    
    // remove the temp file.
    unlink(m3u8_file.c_str());
    
    return ret;
}

int SrsHlsMuxer::_refresh_m3u8(int& fd, string m3u8_file)
{
    int ret = ERROR_SUCCESS;
    
    // no segments, return.
    if (segments.size() == 0) {
        return ret;
    }
    
    int flags = O_CREAT|O_WRONLY|O_TRUNC;
    mode_t mode = S_IRUSR|S_IWUSR|S_IRGRP|S_IWGRP|S_IROTH;
    if ((fd = ::open(m3u8_file.c_str(), flags, mode)) < 0) {
        ret = ERROR_HLS_OPEN_FAILED;
        srs_error("open m3u8 file %s failed. ret=%d", m3u8_file.c_str(), ret);
        return ret;
    }
    srs_info("open m3u8 file %s success.", m3u8_file.c_str());
    
    // #EXTM3U\n#EXT-X-VERSION:3\n
    char header[] = {
        // #EXTM3U\n
        0x23, 0x45, 0x58, 0x54, 0x4d, 0x33, 0x55, 0xa, 
        // #EXT-X-VERSION:3\n
        0x23, 0x45, 0x58, 0x54, 0x2d, 0x58, 0x2d, 0x56, 0x45, 0x52, 
        0x53, 0x49, 0x4f, 0x4e, 0x3a, 0x33, 0xa,
        // #EXT-X-ALLOW-CACHE:NO
        0x23, 0x45, 0x58, 0x54, 0x2d, 0x58, 0x2d, 0x41, 0x4c, 0x4c, 
        0x4f, 0x57, 0x2d, 0x43, 0x41, 0x43, 0x48, 0x45, 0x3a, 0x4e, 0x4f, 0x0a
    };
    if (::write(fd, header, sizeof(header)) != sizeof(header)) {
        ret = ERROR_HLS_WRITE_FAILED;
        srs_error("write m3u8 header failed. ret=%d", ret);
        return ret;
    }
    srs_verbose("write m3u8 header success.");
    
    // #EXT-X-MEDIA-SEQUENCE:4294967295\n
    SrsHlsSegment* first = *segments.begin();
    char sequence[34] = {};
    int len = snprintf(sequence, sizeof(sequence), "#EXT-X-MEDIA-SEQUENCE:%d\n", first->sequence_no);
    if (::write(fd, sequence, len) != len) {
        ret = ERROR_HLS_WRITE_FAILED;
        srs_error("write m3u8 sequence failed. ret=%d", ret);
        return ret;
    }
    srs_verbose("write m3u8 sequence success.");
    
    // #EXT-X-TARGETDURATION:4294967295\n
    int target_duration = 0;
    std::vector<SrsHlsSegment*>::iterator it;
    for (it = segments.begin(); it != segments.end(); ++it) {
        SrsHlsSegment* segment = *it;
        target_duration = srs_max(target_duration, (int)segment->duration);
    }
    // TODO: maybe need to take an around value
    target_duration += 1;
    char duration[34]; // 23+10+1
    len = snprintf(duration, sizeof(duration), "#EXT-X-TARGETDURATION:%d\n", target_duration);
    if (::write(fd, duration, len) != len) {
        ret = ERROR_HLS_WRITE_FAILED;
        srs_error("write m3u8 duration failed. ret=%d", ret);
        return ret;
    }
    srs_verbose("write m3u8 duration success.");
    
    // write all segments
    for (it = segments.begin(); it != segments.end(); ++it) {
        SrsHlsSegment* segment = *it;
        
        if (segment->is_sequence_header) {
            // #EXT-X-DISCONTINUITY\n
            char ext_discon[22]; // 21+1
            len = snprintf(ext_discon, sizeof(ext_discon), "#EXT-X-DISCONTINUITY\n");
            if (::write(fd, ext_discon, len) != len) {
                ret = ERROR_HLS_WRITE_FAILED;
                srs_error("write m3u8 segment discontinuity failed. ret=%d", ret);
                return ret;
            }
            srs_verbose("write m3u8 segment discontinuity success.");
        }
        
        // "#EXTINF:4294967295.208,\n"
        char ext_info[25]; // 14+10+1
        len = snprintf(ext_info, sizeof(ext_info), "#EXTINF:%.3f\n", segment->duration);
        if (::write(fd, ext_info, len) != len) {
            ret = ERROR_HLS_WRITE_FAILED;
            srs_error("write m3u8 segment info failed. ret=%d", ret);
            return ret;
        }
        srs_verbose("write m3u8 segment info success.");
        
        // file name
        std::string filename = segment->uri;
        filename += "\n";
        if (::write(fd, filename.c_str(), filename.length()) != (int)filename.length()) {
            ret = ERROR_HLS_WRITE_FAILED;
            srs_error("write m3u8 segment uri failed. ret=%d", ret);
            return ret;
        }
        srs_verbose("write m3u8 segment uri success.");
    }
    srs_info("write m3u8 %s success.", m3u8_file.c_str());
    
    return ret;
}

int SrsHlsMuxer::create_dir()
{
    int ret = ERROR_SUCCESS;
    
    std::string app_dir = hls_path;
    app_dir += "/";
    app_dir += app;
    
    // TODO: cleanup the dir when startup.

    mode_t mode = S_IRUSR|S_IWUSR|S_IXUSR|S_IRGRP|S_IWGRP|S_IXGRP|S_IROTH|S_IXOTH;
    if (::mkdir(app_dir.c_str(), mode) < 0) {
        if (errno != EEXIST) {
            ret = ERROR_HLS_CREATE_DIR;
            srs_error("create app dir %s failed. ret=%d", app_dir.c_str(), ret);
            return ret;
        }
    }
    srs_info("create app dir %s success.", app_dir.c_str());

    return ret;
}

SrsHlsCache::SrsHlsCache()
{
    aac_jitter = new SrsHlsAacJitter();
    
    ab = new SrsBuffer();
    vb = new SrsBuffer();
    
    af = new SrsMpegtsFrame();
    vf = new SrsMpegtsFrame();
}

SrsHlsCache::~SrsHlsCache()
{
    srs_freep(aac_jitter);
    
    ab->erase(ab->length());
    vb->erase(vb->length());
    
    srs_freep(ab);
    srs_freep(vb);
    
    srs_freep(af);
    srs_freep(vf);
}

int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment_start_dts)
{
    int ret = ERROR_SUCCESS;

    std::string vhost = req->vhost;
    std::string stream = req->stream;
    std::string app = req->app;
    
    int hls_fragment = (int)_srs_config->get_hls_fragment(vhost);
    int hls_window = (int)_srs_config->get_hls_window(vhost);
    
    // get the hls path config
    std::string hls_path = _srs_config->get_hls_path(vhost);
    
    // TODO: FIXME: support load exists m3u8, to continue publish stream.
    // for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
    
    // open muxer
    if ((ret = muxer->update_config(app, stream, hls_path, hls_fragment, hls_window)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer update config failed. ret=%d", ret);
        return ret;
    }
    
    if ((ret = muxer->segment_open(segment_start_dts)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer open segment failed. ret=%d", ret);
        return ret;
    }
    
    return ret;
}

int SrsHlsCache::on_unpublish(SrsHlsMuxer* muxer)
{
    int ret = ERROR_SUCCESS;
    
    if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
        return ret;
    }
    
    if ((ret = muxer->segment_close("unpublish")) != ERROR_SUCCESS) {
        return ret;
    }
    
    return ret;
}

int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
{
    // TODO: support discontinuity for the same stream
    // currently we reap and insert discontinity when encoder republish,
    // but actually, event when stream is not republish, the 
    // sequence header may change, for example,
    // ffmpeg ingest a external rtmp stream and push to srs,
    // when the sequence header changed, the stream is not republish.
    return muxer->on_sequence_header();
}
    
int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
{
    int ret = ERROR_SUCCESS;
    
    // start buffer, set the af
    if (ab->length() == 0) {
        pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
        
        af->dts = af->pts = audio_buffer_start_pts = pts;
        af->pid = TS_AUDIO_PID;
        af->sid = TS_AUDIO_AAC;
    } else {
        aac_jitter->on_buffer_continue();
    }
    
    // write audio to cache.
    if ((ret = cache_audio(codec, sample)) != ERROR_SUCCESS) {
        return ret;
    }
    
    // flush if buffer exceed max size.
    if (ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
        if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
            return ret;
        }
    }
    // TODO: config it.
    // in ms, audio delay to flush the audios.
    int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
    // flush if audio delay exceed
    if (pts - audio_buffer_start_pts > audio_delay * 90) {
        if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
            return ret;
        }
    }
    
    // reap when current source is pure audio.
    // it maybe changed when stream info changed,
    // for example, pure audio when start, audio/video when publishing,
    // pure audio again for audio disabled.
    // so we reap event when the audio incoming when segment overflow.
    // @see https://github.com/simple-rtmp-server/srs/issues/151
    // we use absolutely overflow of segment to make jwplayer/ffplay happy
    // @see https://github.com/simple-rtmp-server/srs/issues/151#issuecomment-71155184
    if (muxer->is_segment_absolutely_overflow()) {
        if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
            return ret;
        }
    }
    
    return ret;
}
    
int SrsHlsCache::write_video(
    SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t dts, SrsCodecSample* sample)
{
    int ret = ERROR_SUCCESS;
    
    // write video to cache.
    if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
        return ret;
    }
    
    vf->dts = dts;
    vf->pts = vf->dts + sample->cts * 90;
    vf->pid = TS_VIDEO_PID;
    vf->sid = TS_VIDEO_AVC;
    vf->key = sample->frame_type == SrsCodecVideoAVCFrameKeyFrame;
    
    // new segment when:
    // 1. base on gop.
    // 2. some gops duration overflow.
    if (vf->key && muxer->is_segment_overflow()) {
        if ((ret = reap_segment("video", muxer, vf->dts)) != ERROR_SUCCESS) {
            return ret;
        }
    }
    
    // flush video when got one
    if ((ret = muxer->flush_video(af, ab, vf, vb)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer flush video failed. ret=%d", ret);
        return ret;
    }
    
    return ret;
}

int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segment_start_dts)
{
    int ret = ERROR_SUCCESS;

    if ((ret = muxer->segment_close(log_desc)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer close segment failed. ret=%d", ret);
        return ret;
    }
    
    if ((ret = muxer->segment_open(segment_start_dts)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer open segment failed. ret=%d", ret);
        return ret;
    }

    // TODO: flush audio before or after segment?
    // segment open, flush the audio.
    // @see: ngx_rtmp_hls_open_fragment
    /* start fragment with audio to make iPhone happy */
    if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
        srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
        return ret;
    }
    
    return ret;
}

int SrsHlsCache::cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample)
{
    int ret = ERROR_SUCCESS;
    
    for (int i = 0; i < sample->nb_sample_units; i++) {
        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
        int32_t size = sample_unit->size;
        
        if (!sample_unit->bytes || size <= 0 || size > 0x1fff) {
            ret = ERROR_HLS_AAC_FRAME_LENGTH;
            srs_error("invalid aac frame length=%d, ret=%d", size, ret);
            return ret;
        }
        
        // the frame length is the AAC raw data plus the adts header size.
        int32_t frame_length = size + 7;
        
        // AAC-ADTS
        // 6.2 Audio Data Transport Stream, ADTS
        // in aac-iso-13818-7.pdf, page 26.
        // fixed 7bytes header
        static u_int8_t adts_header[7] = {0xff, 0xf9, 0x00, 0x00, 0x00, 0x0f, 0xfc};
        /*
        // adts_fixed_header
        // 2B, 16bits
        int16_t syncword; //12bits, '1111 1111 1111'
        int8_t ID; //1bit, '0'
        int8_t layer; //2bits, '00'
        int8_t protection_absent; //1bit, can be '1'
        // 12bits
        int8_t profile; //2bit, 7.1 Profiles, page 40
        TSAacSampleFrequency sampling_frequency_index; //4bits, Table 35, page 46
        int8_t private_bit; //1bit, can be '0'
        int8_t channel_configuration; //3bits, Table 8
        int8_t original_or_copy; //1bit, can be '0'
        int8_t home; //1bit, can be '0'
        
        // adts_variable_header
        // 28bits
        int8_t copyright_identification_bit; //1bit, can be '0'
        int8_t copyright_identification_start; //1bit, can be '0'
        int16_t frame_length; //13bits
        int16_t adts_buffer_fullness; //11bits, 7FF signals that the bitstream is a variable rate bitstream.
        int8_t number_of_raw_data_blocks_in_frame; //2bits, 0 indicating 1 raw_data_block()
        */
        // profile, 2bits
        adts_header[2] = (codec->aac_profile << 6) & 0xc0;
        // sampling_frequency_index 4bits
        adts_header[2] |= (codec->aac_sample_rate << 2) & 0x3c;
        // channel_configuration 3bits
        adts_header[2] |= (codec->aac_channels >> 2) & 0x01;
        adts_header[3] = (codec->aac_channels << 6) & 0xc0;
        // frame_length 13bits
        adts_header[3] |= (frame_length >> 11) & 0x03;
        adts_header[4] = (frame_length >> 3) & 0xff;
        adts_header[5] = ((frame_length << 5) & 0xe0);
        // adts_buffer_fullness; //11bits
        adts_header[5] |= 0x1f;

        // copy to audio buffer
        ab->append((const char*)adts_header, sizeof(adts_header));
        ab->append(sample_unit->bytes, sample_unit->size);
    }
    
    return ret;
}

int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
{
    int ret = ERROR_SUCCESS;
    
    // for type1/5/6, insert aud packet.
    static u_int8_t aud_nal[] = { 0x00, 0x00, 0x00, 0x01, 0x09, 0xf0 };
    
    bool sps_pps_sent = false;
    bool aud_sent = false;
    /**
    * a ts sample is format as:
    * 00 00 00 01 // header
    *       xxxxxxx // data bytes
    * 00 00 01 // continue header
    *       xxxxxxx // data bytes.
    * so, for each sample, we append header in aud_nal, then appends the bytes in sample.
    */
    for (int i = 0; i < sample->nb_sample_units; i++) {
        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
        int32_t size = sample_unit->size;
        
        if (!sample_unit->bytes || size <= 0) {
            ret = ERROR_HLS_AVC_SAMPLE_SIZE;
            srs_error("invalid avc sample length=%d, ret=%d", size, ret);
            return ret;
        }
        
        /**
        * step 1:
        * first, before each "real" sample, 
        * we add some packets according to the nal_unit_type,
        * for example, when got nal_unit_type=5, insert SPS/PPS before sample.
        */
        
        // 5bits, 7.3.1 NAL unit syntax, 
        // H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
        u_int8_t nal_unit_type;
        nal_unit_type = *sample_unit->bytes;
        nal_unit_type &= 0x1f;
        
        // @see: ngx_rtmp_hls_video
        // Table 7-1 – NAL unit type codes, page 61
        // 1: Coded slice
        if (nal_unit_type == 1) {
            sps_pps_sent = false;
        }
        
        // 6: Supplemental enhancement information (SEI) sei_rbsp( ), page 61
        // @see: ngx_rtmp_hls_append_aud
        if (!aud_sent) {
            // @remark, when got type 9, we donot send aud_nal, but it will make 
            //      ios unhappy, so we remove it.
            // @see https://github.com/simple-rtmp-server/srs/issues/281
            /*if (nal_unit_type == 9) {
                aud_sent = true;
            }*/
            
            if (nal_unit_type == 1 || nal_unit_type == 5 || nal_unit_type == 6) {
                // for type 6, append a aud with type 9.
                vb->append((const char*)aud_nal, sizeof(aud_nal));
                aud_sent = true;
            }
        }
        
        // 5: Coded slice of an IDR picture.
        // insert sps/pps before IDR or key frame is ok.
        if (nal_unit_type == 5 && !sps_pps_sent) {
            sps_pps_sent = true;
            
            // @see: ngx_rtmp_hls_append_sps_pps
            if (codec->sequenceParameterSetLength > 0) {
                // AnnexB prefix, for sps always 4 bytes header
                vb->append((const char*)aud_nal, 4);
                // sps
                vb->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
            }
            if (codec->pictureParameterSetLength > 0) {
                // AnnexB prefix, for pps always 4 bytes header
                vb->append((const char*)aud_nal, 4);
                // pps
                vb->append(codec->pictureParameterSetNALUnit, codec->pictureParameterSetLength);
            }
        }
        
        // 7-9, ignore, @see: ngx_rtmp_hls_video
        if (nal_unit_type >= 7 && nal_unit_type <= 9) {
            continue;
        }
        
        /**
        * step 2:
        * output the "real" sample, in buf.
        * when we output some special assist packets according to nal_unit_type
        */
        
        // sample start prefix, '00 00 00 01' or '00 00 01'
        u_int8_t* p = aud_nal + 1;
        u_int8_t* end = p + 3;
        
        // first AnnexB prefix is long (4 bytes)
        if (vb->length() == 0) {
            p = aud_nal;
        }
        vb->append((const char*)p, end - p);
        
        // sample data
        vb->append(sample_unit->bytes, sample_unit->size);
    }
    
    return ret;
}

SrsHls::SrsHls(SrsSource* _source)
{
    hls_enabled = false;
    
    source = _source;
    codec = new SrsAvcAacCodec();
    sample = new SrsCodecSample();
    jitter = new SrsRtmpJitter();
    
    muxer = new SrsHlsMuxer();
    hls_cache = new SrsHlsCache();

    pithy_print = new SrsPithyPrint(SRS_CONSTS_STAGE_HLS);
    stream_dts = 0;
}

SrsHls::~SrsHls()
{
    srs_freep(codec);
    srs_freep(sample);
    srs_freep(jitter);
    
    srs_freep(muxer);
    srs_freep(hls_cache);
    
    srs_freep(pithy_print);
}

int SrsHls::on_publish(SrsRequest* req)
{
    int ret = ERROR_SUCCESS;
    
    // support multiple publish.
    if (hls_enabled) {
        return ret;
    }
    
    std::string vhost = req->vhost;
    if (!_srs_config->get_hls_enabled(vhost)) {
        return ret;
    }
    
    if ((ret = hls_cache->on_publish(muxer, req, stream_dts)) != ERROR_SUCCESS) {
        return ret;
    }
    
    // if enabled, open the muxer.
    hls_enabled = true;
    
    // notice the source to get the cached sequence header.
    // when reload to start hls, hls will never get the sequence header in stream,
    // use the SrsSource.on_hls_start to push the sequence header to HLS.
    if ((ret = source->on_hls_start()) != ERROR_SUCCESS) {
        srs_error("callback source hls start failed. ret=%d", ret);
        return ret;
    }

    return ret;
}

void SrsHls::on_unpublish()
{
    int ret = ERROR_SUCCESS;
    
    // support multiple unpublish.
    if (!hls_enabled) {
        return;
    }

    if ((ret = hls_cache->on_unpublish(muxer)) != ERROR_SUCCESS) {
        srs_error("ignore m3u8 muxer flush/close audio failed. ret=%d", ret);
    }
    
    hls_enabled = false;
}

int SrsHls::on_meta_data(SrsAmf0Object* metadata)
{
    int ret = ERROR_SUCCESS;

    if (!metadata) {
        srs_trace("no metadata persent, hls ignored it.");
        return ret;
    }
    
    if (metadata->count() <= 0) {
        srs_trace("no metadata persent, hls ignored it.");
        return ret;
    }
    
    if ((ret = codec->metadata_demux(metadata)) != ERROR_SUCCESS) {
        return ret;
    }
    
    return ret;
}

int SrsHls::on_audio(SrsSharedPtrMessage* audio)
{
    int ret = ERROR_SUCCESS;
    
    SrsAutoFree(SrsSharedPtrMessage, audio);
    
    if (!hls_enabled) {
        return ret;
    }
    
    sample->clear();
    if ((ret = codec->audio_aac_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
        srs_error("hls codec demux audio failed. ret=%d", ret);
        return ret;
    }
    
    if (codec->audio_codec_id != SrsCodecAudioAAC) {
        return ret;
    }
    
    // ignore sequence header
    if (sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
        return hls_cache->on_sequence_header(muxer);
    }
    
    if ((ret = jitter->correct(audio, 0, 0, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
        srs_error("rtmp jitter correct audio failed. ret=%d", ret);
        return ret;
    }
    
    // the pts calc from rtmp/flv header.
    int64_t pts = audio->header.timestamp * 90;
    
    // for pure audio, we need to update the stream dts also.
    stream_dts = pts;
    
    if ((ret = hls_cache->write_audio(codec, muxer, pts, sample)) != ERROR_SUCCESS) {
        srs_error("hls cache write audio failed. ret=%d", ret);
        return ret;
    }
    
    return ret;
}

int SrsHls::on_video(SrsSharedPtrMessage* video)
{
    int ret = ERROR_SUCCESS;
    
    SrsAutoFree(SrsSharedPtrMessage, video);
    
    if (!hls_enabled) {
        return ret;
    }
    
    sample->clear();
    if ((ret = codec->video_avc_demux(video->payload, video->size, sample)) != ERROR_SUCCESS) {
        srs_error("hls codec demux video failed. ret=%d", ret);
        return ret;
    }
    
    // ignore info frame,
    // @see https://github.com/simple-rtmp-server/srs/issues/288#issuecomment-69863909
    if (sample->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
        return ret;
    }
    
    if (codec->video_codec_id != SrsCodecVideoAVC) {
        return ret;
    }
    
    // ignore sequence header
    if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame
         && sample->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
        return hls_cache->on_sequence_header(muxer);
    }
    
    if ((ret = jitter->correct(video, 0, 0, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
        srs_error("rtmp jitter correct video failed. ret=%d", ret);
        return ret;
    }
    
    int64_t dts = video->header.timestamp * 90;
    stream_dts = dts;
    if ((ret = hls_cache->write_video(codec, muxer, dts, sample)) != ERROR_SUCCESS) {
        srs_error("hls cache write video failed. ret=%d", ret);
        return ret;
    }
    
    hls_mux();
    
    return ret;
}

void SrsHls::hls_mux()
{
    // reportable
    if (pithy_print->can_print()) {
        // the run time is not equals to stream time,
        // @see: https://github.com/simple-rtmp-server/srs/issues/81#issuecomment-48100994
        // it's ok.
        srs_trace("-> "SRS_CONSTS_LOG_HLS
            " time=%"PRId64", stream dts=%"PRId64"(%"PRId64"ms), sequence_no=%d", 
            pithy_print->age(), stream_dts, stream_dts / 90, muxer->sequence_no());
    }
    
    pithy_print->elapse();
}

#endif