/**
 * The MIT License (MIT)
 *
 * Copyright (c) 2013-2017 OSSRS(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 <stdio.h>
#include <stdlib.h>
#include <unistd.h>

// for open audio raw file.
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>

#include "../../objs/include/srs_librtmp.h"

// https://github.com/ossrs/srs/issues/212#issuecomment-64145910
int read_audio_frame(char* data, int size, char** pp, char** frame, int* frame_size)
{
    char* p = *pp;
    
    // @remark, for this demo, to publish aac raw file to SRS,
    // we search the adts frame from the buffer which cached the aac data.
    // please get aac adts raw data from device, it always a encoded frame.
    if (!srs_aac_is_adts(p, size - (p - data))) {
        srs_human_trace("aac adts raw data invalid.");
        return -1;
    }
    
    // @see srs_audio_write_raw_frame
    // each frame prefixed aac adts header, '1111 1111 1111'B, that is 0xFFF.,
    // for instance, frame = FF F1 5C 80 13 A0 FC 00 D0 33 83 E8 5B
    *frame = p;
    // skip some data.
    // @remark, user donot need to do this.
    p += srs_aac_adts_frame_size(p, size - (p - data));
    
    *pp = p;
    *frame_size = p - *frame;
    if (*frame_size <= 0) {
        srs_human_trace("aac adts raw data invalid.");
        return -1;
    }
    
    return 0;
}

int main(int argc, char** argv)
{
    printf("publish raw audio as rtmp stream to server like FMLE/FFMPEG/Encoder\n");
    printf("SRS(ossrs) client librtmp library.\n");
    printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
    
    if (argc <= 2) {
        printf("Usage: %s <audio_raw_file> <rtmp_publish_url>\n", argv[0]);
        printf("     audio_raw_file: the audio raw steam file.\n");
        printf("     rtmp_publish_url: the rtmp publish url.\n");
        printf("For example:\n");
        printf("     %s ./audio.raw.aac rtmp://127.0.0.1:1935/live/livestream\n", argv[0]);
        printf("Where the file: http://winlinvip.github.io/srs.release/3rdparty/audio.raw.aac\n");
        printf("See: https://github.com/ossrs/srs/issues/212\n");
        exit(-1);
    }
    
    const char* raw_file = argv[1];
    const char* rtmp_url = argv[2];
    srs_human_trace("raw_file=%s, rtmp_url=%s", raw_file, rtmp_url);
    
    // open file
    int raw_fd = open(raw_file, O_RDONLY);
    if (raw_fd < 0) {
        srs_human_trace("open audio raw file %s failed.", raw_file);
        goto rtmp_destroy;
    }
    
    off_t file_size = lseek(raw_fd, 0, SEEK_END);
    if (file_size <= 0) {
        srs_human_trace("audio raw file %s empty.", raw_file);
        goto rtmp_destroy;
    }
    srs_human_trace("read entirely audio raw file, size=%dKB", (int)(file_size / 1024));
    
    char* audio_raw = (char*)malloc(file_size);
    if (!audio_raw) {
        srs_human_trace("alloc raw buffer failed for file %s.", raw_file);
        goto rtmp_destroy;
    }
    
    lseek(raw_fd, 0, SEEK_SET);
    ssize_t nb_read = 0;
    if ((nb_read = read(raw_fd, audio_raw, file_size)) != file_size) {
        srs_human_trace("buffer %s failed, expect=%dKB, actual=%dKB.",
            raw_file, (int)(file_size / 1024), (int)(nb_read / 1024));
        goto rtmp_destroy;
    }
    
    // connect rtmp context
    srs_rtmp_t rtmp = srs_rtmp_create(rtmp_url);
    
    if (srs_rtmp_handshake(rtmp) != 0) {
        srs_human_trace("simple handshake failed.");
        goto rtmp_destroy;
    }
    srs_human_trace("simple handshake success");
    
    if (srs_rtmp_connect_app(rtmp) != 0) {
        srs_human_trace("connect vhost/app failed.");
        goto rtmp_destroy;
    }
    srs_human_trace("connect vhost/app success");
    
    if (srs_rtmp_publish_stream(rtmp) != 0) {
        srs_human_trace("publish stream failed.");
        goto rtmp_destroy;
    }
    srs_human_trace("publish stream success");
    
    uint32_t timestamp = 0;
    uint32_t time_delta = 45;
    // @remark, to decode the file.
    char* p = audio_raw;
    for (;p < audio_raw + file_size;) {
        // @remark, read a frame from file buffer.
        char* data = NULL;
        int size = 0;
        if (read_audio_frame(audio_raw, file_size, &p, &data, &size) < 0) {
            srs_human_trace("read a frame from file buffer failed.");
            goto rtmp_destroy;
        }
        
        // 0 = Linear PCM, platform endian
        // 1 = ADPCM
        // 2 = MP3
        // 7 = G.711 A-law logarithmic PCM
        // 8 = G.711 mu-law logarithmic PCM
        // 10 = AAC
        // 11 = Speex
        char sound_format = 10;
        // 2 = 22 kHz
        char sound_rate = 2;
        // 1 = 16-bit samples
        char sound_size = 1;
        // 1 = Stereo sound
        char sound_type = 1;
        
        timestamp += time_delta;
        
        int ret = 0;
        if ((ret = srs_audio_write_raw_frame(rtmp, sound_format, sound_rate, sound_size, sound_type, data, size, timestamp)) != 0) {
            srs_human_trace("send audio raw data failed. ret=%d", ret);
            goto rtmp_destroy;
        }
        
        srs_human_trace("sent packet: type=%s, time=%d, size=%d, codec=%d, rate=%d, sample=%d, channel=%d",
            srs_human_flv_tag_type2string(SRS_RTMP_TYPE_AUDIO), timestamp, size, sound_format, sound_rate, sound_size,
            sound_type);
        
        // @remark, when use encode device, it not need to sleep.
        usleep(1000 * time_delta);
    }
    
rtmp_destroy:
    srs_rtmp_destroy(rtmp);
    close(raw_fd);
    free(audio_raw);
    
    return 0;
}