gpac录制h264/h265 + aac 的mp4文件
gpac录制h264/h265 + aac 的mp4文件
- gpac在windows端编译
- 参考代码
- 贴代码
- 备注
- 如果有用，送点下载分给我
gpac在windows端编译
源码下载
我下的最新0.8release，以下代码基于这版本。
编译很简单在代码包build目录下有vs的工程文件，选mp4box mini那个，代码生成模式改成mt编译
懒得编译送我点分，我提供
参考代码
https://github.com/Tvirus/gpac-interface
非常感谢这位大佬，可能版本的问题，改动较大，直接使用问题很大。对比源码修改不少，并添加了h264支持。
贴代码
#ifndef _MP4WRITER_H_
#define _MP4WRITER_H_
#define GPAC_MP4BOX_MINI
#include "gpac/setup.h"
#define MP4_AUDIO_TYPE_INVALID 0
#define MP4_AUDIO_TYPE_AAC_MAIN 1
#define MP4_AUDIO_TYPE_AAC_LC 2
#define MP4_AUDIO_TYPE_AAC_SSR 3
#define MP4_AUDIO_TYPE_AAC_LD 23
#ifdef __cplusplus
extern "C" {
#endif
void* MP4_Init();
s32 MP4_CreatFile(void *pCMP4Writer, char *strFileName);
s32 MP4_InitVideo265(void *pCMP4Writer, u32 TimeScale);
s32 MP4_Write265Sample(void *pCMP4Writer, u8 *pData, u32 Size, u64 TimeStamp);
s32 MP4_InitVideo264(void *pCMP4Writer, u32 TimeScale);
s32 MP4_Write264Sample(void *pCMP4Writer, u8 *pData, u32 Size, u64 TimeStamp);
s32 MP4_InitAudioAAC(void *pCMP4Writer, u8 AudioType, u32 SampleRate, u8 Channel, u32 TimeScale);
s32 MP4_WriteAACSample(void *pCMP4Writer, u8 *pData, u32 Size, u64 TimeStamp);
void MP4_CloseFile(void *pCMP4Writer);
void MP4_Exit(void *pCMP4Writer);
#ifdef __cplusplus
}
#endif
#endif
#include "MP4Writer.h"
#include 备注
MP4_InitAudioAAC那个audiotype指的是aac的profile，lc就是2。
GF_M4A_AAC_MAIN = 1,
GF_M4A_AAC_LC = 2,
GF_M4A_AAC_SSR = 3,
GF_M4A_AAC_LTP = 4,
GF_M4A_AAC_SBR = 5,
GF_M4A_AAC_SCALABLE = 6,
GF_M4A_TWINVQ = 7,
GF_M4A_CELP = 8,
GF_M4A_HVXC = 9,
GF_M4A_TTSI = 12,
GF_M4A_MAIN_SYNTHETIC = 13,
GF_M4A_WAVETABLE_SYNTHESIS = 14,
GF_M4A_GENERAL_MIDI = 15,
GF_M4A_ALGO_SYNTH_AUDIO_FX = 16,
GF_M4A_ER_AAC_LC = 17,
GF_M4A_ER_AAC_LTP = 19,
GF_M4A_ER_AAC_SCALABLE = 20,
GF_M4A_ER_TWINVQ = 21,
GF_M4A_ER_BSAC = 22,
GF_M4A_ER_AAC_LD = 23,
GF_M4A_ER_CELP = 24,
GF_M4A_ER_HVXC = 25,
GF_M4A_ER_HILN = 26,
GF_M4A_ER_PARAMETRIC = 27,
GF_M4A_SSC = 28,
GF_M4A_AAC_PS = 29,
GF_M4A_LAYER1 = 32,
GF_M4A_LAYER2 = 33,
GF_M4A_LAYER3 = 34,
GF_M4A_DST = 35,
GF_M4A_ALS = 36
填aac数据时记得除掉adts头
可以用我这个来试试。
static int AdtsDemux(unsigned char * data, unsigned int size, unsigned char** raw, int* raw_size)
{
int ret = 1;
if (size < 7) {
ttf_human_trace("adts: demux size too small");
return 0;
}
unsigned char* p = data;
unsigned char* pend = data + size;
//unsigned char* startp = 0;
while (p < pend) {
// decode the ADTS.
// @see aac-iso-13818-7.pdf, page 26
// 6.2 Audio Data Transport Stream, ADTS
// @see https://github.com/ossrs/srs/issues/212#issuecomment-64145885
// byte_alignment()
// adts_fixed_header:
// 12bits syncword,
// 16bits left.
// adts_variable_header:
// 28bits
// 12+16+28=56bits
// adts_error_check:
// 16bits if protection_absent
// 56+16=72bits
// if protection_absent:
// require(7bytes)=56bits
// else
// require(9bytes)=72bits
//startp = p;
// for aac, the frame must be ADTS format.
/*if (p[0] != 0xff || (p[1] & 0xf0) != 0xf0) {
ttf_human_trace("adts: not this format.");
return 0;
}*/
// syncword 12 bslbf
p++;
// 4bits left.
// adts_fixed_header(), 1.A.2.2.1 Fixed Header of ADTS
// ID 1 bslbf
// layer 2 uimsbf
// protection_absent 1 bslbf
int8_t pav = (*p++ & 0x0f);
//int8_t id = (pav >> 3) & 0x01;
/*int8_t layer = (pav >> 1) & 0x03;*/
int8_t protection_absent = pav & 0x01;
/**
* ID: MPEG identifier, set to '1' if the audio data in the ADTS stream are MPEG-2 AAC (See ISO/IEC 13818-7)
* and set to '0' if the audio data are MPEG-4. See also ISO/IEC 11172-3, subclause 2.4.2.3.
*/
//if (id != 0x01) {
// //ttf_human_trace("adts: id must be 1(aac), actual 0(mp4a).");
// // well, some system always use 0, but actually is aac format.
// // for example, houjian vod ts always set the aac id to 0, actually 1.
// // we just ignore it, and alwyas use 1(aac) to demux.
// id = 0x01;
//}
//else {
// //ttf_human_trace("adts: id must be 1(aac), actual 1(mp4a).");
//}
//int16_t sfiv = (*p << 8) | (*(p + 1));
p += 2;
// profile 2 uimsbf
// sampling_frequency_index 4 uimsbf
// private_bit 1 bslbf
// channel_configuration 3 uimsbf
// original/copy 1 bslbf
// home 1 bslbf
//int8_t profile = (sfiv >> 14) & 0x03;
//int8_t sampling_frequency_index = (sfiv >> 10) & 0x0f;
/*int8_t private_bit = (sfiv >> 9) & 0x01;*/
//int8_t channel_configuration = (sfiv >> 6) & 0x07;
/*int8_t original = (sfiv >> 5) & 0x01;*/
/*int8_t home = (sfiv >> 4) & 0x01;*/
//int8_t Emphasis; @remark, Emphasis is removed, @see https://github.com/ossrs/srs/issues/212#issuecomment-64154736
// 4bits left.
// adts_variable_header(), 1.A.2.2.2 Variable Header of ADTS
// copyright_identification_bit 1 bslbf
// copyright_identification_start 1 bslbf
/*int8_t fh_copyright_identification_bit = (fh1 >> 3) & 0x01;*/
/*int8_t fh_copyright_identification_start = (fh1 >> 2) & 0x01;*/
// frame_length 13 bslbf: Length of the frame including headers and error_check in bytes.
// use the left 2bits as the 13 and 12 bit,
// the frame_length is 13bits, so we move 13-2=11.
//int16_t frame_length = (sfiv << 11) & 0x1800;
//int32_t abfv = ((*p) << 16)
// | ((*(p + 1)) << 8)
// | (*(p + 2));
p += 3;
// frame_length 13 bslbf: consume the first 13-2=11bits
// the fh2 is 24bits, so we move right 24-11=13.
//frame_length |= (abfv >> 13) & 0x07ff;
// adts_buffer_fullness 11 bslbf
/*int16_t fh_adts_buffer_fullness = (abfv >> 2) & 0x7ff;*/
// number_of_raw_data_blocks_in_frame 2 uimsbf
/*int16_t number_of_raw_data_blocks_in_frame = abfv & 0x03;*/
// adts_error_check(), 1.A.2.2.3 Error detection
if (!protection_absent) {
if (size < 9) {
ttf_human_trace("adts: protection_absent disappare.");
return 0;
}
// crc_check 16 Rpchof
/*int16_t crc_check = */p += 2;
}
// TODO: check the sampling_frequency_index
// TODO: check the channel_configuration
// raw_data_blocks
/*int adts_header_size = p - startp;
int raw_data_size = frame_length - adts_header_size;
if (raw_data_size > pend - p) {
ttf_human_trace("adts: raw data size too little.");
return 0;
}*/
//adts_codec_.protection_absent = protection_absent;
//adts_codec_.aac_object = srs_codec_aac_ts2rtmp((SrsAacProfile)profile);
//adts_codec_.sampling_frequency_index = sampling_frequency_index;
//adts_codec_.channel_configuration = channel_configuration;
//adts_codec_.frame_length = frame_length;
//// @see srs_audio_write_raw_frame().
//// TODO: FIXME: maybe need to resample audio.
//adts_codec_.sound_format = 10; // AAC
//if (sampling_frequency_index <= 0x0c && sampling_frequency_index > 0x0a) {
// adts_codec_.sound_rate = SrsCodecAudioSampleRate5512;
//}
//else if (sampling_frequency_index <= 0x0a && sampling_frequency_index > 0x07) {
// adts_codec_.sound_rate = SrsCodecAudioSampleRate11025;
//}
//else if (sampling_frequency_index <= 0x07 && sampling_frequency_index > 0x04) {
// adts_codec_.sound_rate = SrsCodecAudioSampleRate22050;
//}
//else if (sampling_frequency_index <= 0x04) {
// adts_codec_.sound_rate = SrsCodecAudioSampleRate44100;
//}
//else {
// adts_codec_.sound_rate = SrsCodecAudioSampleRate44100;
// //srs_warn("adts invalid sample rate for flv, rate=%#x", sampling_frequency_index);
//}
//adts_codec_.sound_type = srs_max(0, srs_min(1, channel_configuration - 1));
//// TODO: FIXME: finger it out the sound size by adts.
//adts_codec_.sound_size = 1; // 0(8bits) or 1(16bits).
// frame data.
*raw = p;
*raw_size = pend - p;
break;
}
return ret;
}