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ea85ad2e 1 parent d22e4e86

for #293, #250, move the ts codec to kernel ts.

内嵌 并排对比
正在显示 7 个修改的文件 包含 829 行增加 和 800 行删除
... ... @@ -50,6 +50,7 @@ using namespace std;
#include <srs_kernel_avc.hpp>
#include <srs_kernel_file.hpp>
#include <srs_rtmp_buffer.hpp>
#include <srs_kernel_ts.hpp>
// drop the segment when duration of ts too small.
#define SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS 100
... ...
... ... @@ -33,6 +33,9 @@ using namespace std;
#include <srs_kernel_log.hpp>
#include <srs_app_config.hpp>
// Transport Stream packets are 188 bytes in length.
#define TS_PACKET_SIZE 188
#ifdef SRS_AUTO_STREAM_CASTER
SrsMpegtsOverUdp::SrsMpegtsOverUdp(SrsConfDirective* c)
... ... @@ -51,10 +54,28 @@ int SrsMpegtsOverUdp::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
std::string peer_ip = inet_ntoa(from->sin_addr);
int peer_port = ntohs(from->sin_port);
// drop ts packet when size not modulus by 188
if (nb_buf < TS_PACKET_SIZE || (nb_buf % TS_PACKET_SIZE) != 0) {
srs_warn("udp: drop %s:%d packet %d bytes", peer_ip.c_str(), peer_port, nb_buf);
return ret;
}
srs_info("udp: got %s:%d packet %d bytes", peer_ip.c_str(), peer_port, nb_buf);
// TODO: FIXME: implements it.
// process each ts packet
for (int i = 0; i < nb_buf; i += TS_PACKET_SIZE) {
char* ts_packet = buf + i;
if ((ret = on_ts_packet(ts_packet)) != ERROR_SUCCESS) {
srs_warn("mpegts: ignore ts packet error. ret=%d", ret);
continue;
}
}
return ret;
}
int SrsMpegtsOverUdp::on_ts_packet(char* ts_packet)
{
int ret = ERROR_SUCCESS;
return ret;
}
... ...
... ... @@ -58,6 +58,11 @@ public:
* @remark user should never use the buf, for it's a shared memory bytes.
*/
virtual int on_udp_packet(sockaddr_in* from, char* buf, int nb_buf);
private:
/**
* when got a ts packet, in size TS_PACKET_SIZE.
*/
virtual int on_ts_packet(char* ts_packet);
};
#endif
... ...
... ... @@ -32,702 +32,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
using namespace std;
// in ms, for HLS aac sync time.
#define SRS_CONF_DEFAULT_AAC_SYNC 100
// @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
// 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
/**
* 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
};
// @see: NGX_RTMP_HLS_DELAY,
// 63000: 700ms, ts_tbn=90000
#define SRS_AUTO_HLS_DELAY 63000
// @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/winlinvip/simple-rtmp-server/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_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, SrsSimpleBuffer* 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
p = write_pcr(p, frame->dts - SRS_AUTO_HLS_DELAY);
}
// 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) {
if (!srs_is_client_gracefully_close(ret)) {
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
// and the pcr maybe negative
// @see https://github.com/winlinvip/simple-rtmp-server/issues/268
int64_t v = srs_max(0, 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;
}
};
SrsMpegtsFrame::SrsMpegtsFrame()
{
pts = dts = 0;
pid = sid = cc = 0;
key = false;
}
SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
{
writer = w;
}
SrsTSMuxer::~SrsTSMuxer()
{
close();
}
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, SrsSimpleBuffer* 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, SrsSimpleBuffer* vb)
{
int ret = ERROR_SUCCESS;
if ((ret = SrsMpegtsWriter::write_frame(writer, vf, vb)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
void SrsTSMuxer::close()
{
writer->close();
}
SrsTsAacJitter::SrsTsAacJitter()
{
base_pts = 0;
nb_samples = 0;
// TODO: config it, 0 means no adjust
sync_ms = SRS_CONF_DEFAULT_AAC_SYNC;
}
SrsTsAacJitter::~SrsTsAacJitter()
{
}
int64_t SrsTsAacJitter::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 SrsTsAacJitter::on_buffer_continue()
{
nb_samples++;
}
SrsTsCache::SrsTsCache()
{
aac_jitter = new SrsTsAacJitter();
ab = new SrsSimpleBuffer();
vb = new SrsSimpleBuffer();
af = new SrsMpegtsFrame();
vf = new SrsMpegtsFrame();
}
SrsTsCache::~SrsTsCache()
{
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 SrsTsCache::cache_audio(SrsAvcAacCodec* codec, 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 = pts;
af->pid = TS_AUDIO_PID;
af->sid = TS_AUDIO_AAC;
} else {
aac_jitter->on_buffer_continue();
}
// write audio to cache.
if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
int SrsTsCache::cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
// write video to cache.
if ((ret = do_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;
return ret;
}
int SrsTsCache::do_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, 0xf1, 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 SrsTsCache::do_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 C 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/winlinvip/simple-rtmp-server/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;
}
SrsCodecSampleUnit::SrsCodecSampleUnit()
{
size = 0;
... ...
... ... @@ -95,109 +95,6 @@ enum SrsCodecAudioSoundType
SrsCodecAudioSoundTypeStereo = 1,
};
// @see: ngx_rtmp_SrsMpegtsFrame_t
class SrsMpegtsFrame
{
public:
int64_t pts;
int64_t dts;
int pid;
int sid;
int cc;
bool key;
SrsMpegtsFrame();
};
/**
* write data from frame(header info) and buffer(data) to ts file.
* it's a simple object wrapper for utility from nginx-rtmp: SrsMpegtsWriter
*/
class SrsTSMuxer
{
private:
SrsFileWriter* writer;
std::string path;
public:
SrsTSMuxer(SrsFileWriter* w);
virtual ~SrsTSMuxer();
public:
virtual int open(std::string _path);
virtual int write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
virtual int write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
virtual void close();
};
/**
* jitter correct for audio,
* the sample rate 44100/32000 will lost precise,
* when mp4/ts(tbn=90000) covert to flv/rtmp(1000),
* so the Hls on ipad or iphone will corrupt,
* @see nginx-rtmp: est_pts
*/
class SrsTsAacJitter
{
private:
int64_t base_pts;
int64_t nb_samples;
int sync_ms;
public:
SrsTsAacJitter();
virtual ~SrsTsAacJitter();
/**
* when buffer start, calc the "correct" pts for ts,
* @param flv_pts, the flv pts calc from flv header timestamp,
* @param sample_rate, the sample rate in format(flv/RTMP packet header).
* @param aac_sample_rate, the sample rate in codec(sequence header).
* @return the calc correct pts.
*/
virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
/**
* when buffer continue, muxer donot write to file,
* the audio buffer continue grow and donot need a pts,
* for the ts audio PES packet only has one pts at the first time.
*/
virtual void on_buffer_continue();
};
/**
* ts stream cache,
* use to cache ts stream.
*
* about the flv tbn problem:
* flv tbn is 1/1000, ts tbn is 1/90000,
* when timestamp convert to flv tbn, it will loose precise,
* so we must gather audio frame together, and recalc the timestamp @see SrsTsAacJitter,
* we use a aac jitter to correct the audio pts.
*/
class SrsTsCache
{
public:
// current frame and buffer
SrsMpegtsFrame* af;
SrsSimpleBuffer* ab;
SrsMpegtsFrame* vf;
SrsSimpleBuffer* vb;
protected:
// time jitter for aac
SrsTsAacJitter* aac_jitter;
public:
SrsTsCache();
virtual ~SrsTsCache();
public:
/**
* write audio to cache
*/
virtual int cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample);
/**
* write video to muxer.
*/
virtual int cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
private:
virtual int do_cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int do_cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
};
/**
* the codec sample unit.
* for h.264 video packet, a NALU is a sample unit.
... ...
... ... @@ -37,6 +37,703 @@ using namespace std;
#include <srs_kernel_file.hpp>
#include <srs_kernel_avc.hpp>
#include <srs_kernel_buffer.hpp>
#include <srs_kernel_utility.hpp>
// in ms, for HLS aac sync time.
#define SRS_CONF_DEFAULT_AAC_SYNC 100
// @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
// 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
/**
* 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
};
// @see: NGX_RTMP_HLS_DELAY,
// 63000: 700ms, ts_tbn=90000
#define SRS_AUTO_HLS_DELAY 63000
// @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/winlinvip/simple-rtmp-server/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_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, SrsSimpleBuffer* 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
p = write_pcr(p, frame->dts - SRS_AUTO_HLS_DELAY);
}
// 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) {
if (!srs_is_client_gracefully_close(ret)) {
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
// and the pcr maybe negative
// @see https://github.com/winlinvip/simple-rtmp-server/issues/268
int64_t v = srs_max(0, 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;
}
};
SrsMpegtsFrame::SrsMpegtsFrame()
{
pts = dts = 0;
pid = sid = cc = 0;
key = false;
}
SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
{
writer = w;
}
SrsTSMuxer::~SrsTSMuxer()
{
close();
}
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, SrsSimpleBuffer* 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, SrsSimpleBuffer* vb)
{
int ret = ERROR_SUCCESS;
if ((ret = SrsMpegtsWriter::write_frame(writer, vf, vb)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
void SrsTSMuxer::close()
{
writer->close();
}
SrsTsAacJitter::SrsTsAacJitter()
{
base_pts = 0;
nb_samples = 0;
// TODO: config it, 0 means no adjust
sync_ms = SRS_CONF_DEFAULT_AAC_SYNC;
}
SrsTsAacJitter::~SrsTsAacJitter()
{
}
int64_t SrsTsAacJitter::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 SrsTsAacJitter::on_buffer_continue()
{
nb_samples++;
}
SrsTsCache::SrsTsCache()
{
aac_jitter = new SrsTsAacJitter();
ab = new SrsSimpleBuffer();
vb = new SrsSimpleBuffer();
af = new SrsMpegtsFrame();
vf = new SrsMpegtsFrame();
}
SrsTsCache::~SrsTsCache()
{
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 SrsTsCache::cache_audio(SrsAvcAacCodec* codec, 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 = pts;
af->pid = TS_AUDIO_PID;
af->sid = TS_AUDIO_AAC;
} else {
aac_jitter->on_buffer_continue();
}
// write audio to cache.
if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
int SrsTsCache::cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
// write video to cache.
if ((ret = do_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;
return ret;
}
int SrsTsCache::do_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, 0xf1, 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 SrsTsCache::do_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 C 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/winlinvip/simple-rtmp-server/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;
}
SrsTsEncoder::SrsTsEncoder()
{
... ...
... ... @@ -37,6 +37,110 @@ class SrsFileWriter;
class SrsFileReader;
class SrsAvcAacCodec;
class SrsCodecSample;
class SrsSimpleBuffer;
// @see: ngx_rtmp_SrsMpegtsFrame_t
class SrsMpegtsFrame
{
public:
int64_t pts;
int64_t dts;
int pid;
int sid;
int cc;
bool key;
SrsMpegtsFrame();
};
/**
* write data from frame(header info) and buffer(data) to ts file.
* it's a simple object wrapper for utility from nginx-rtmp: SrsMpegtsWriter
*/
class SrsTSMuxer
{
private:
SrsFileWriter* writer;
std::string path;
public:
SrsTSMuxer(SrsFileWriter* w);
virtual ~SrsTSMuxer();
public:
virtual int open(std::string _path);
virtual int write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
virtual int write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
virtual void close();
};
/**
* jitter correct for audio,
* the sample rate 44100/32000 will lost precise,
* when mp4/ts(tbn=90000) covert to flv/rtmp(1000),
* so the Hls on ipad or iphone will corrupt,
* @see nginx-rtmp: est_pts
*/
class SrsTsAacJitter
{
private:
int64_t base_pts;
int64_t nb_samples;
int sync_ms;
public:
SrsTsAacJitter();
virtual ~SrsTsAacJitter();
/**
* when buffer start, calc the "correct" pts for ts,
* @param flv_pts, the flv pts calc from flv header timestamp,
* @param sample_rate, the sample rate in format(flv/RTMP packet header).
* @param aac_sample_rate, the sample rate in codec(sequence header).
* @return the calc correct pts.
*/
virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
/**
* when buffer continue, muxer donot write to file,
* the audio buffer continue grow and donot need a pts,
* for the ts audio PES packet only has one pts at the first time.
*/
virtual void on_buffer_continue();
};
/**
* ts stream cache,
* use to cache ts stream.
*
* about the flv tbn problem:
* flv tbn is 1/1000, ts tbn is 1/90000,
* when timestamp convert to flv tbn, it will loose precise,
* so we must gather audio frame together, and recalc the timestamp @see SrsTsAacJitter,
* we use a aac jitter to correct the audio pts.
*/
class SrsTsCache
{
public:
// current frame and buffer
SrsMpegtsFrame* af;
SrsSimpleBuffer* ab;
SrsMpegtsFrame* vf;
SrsSimpleBuffer* vb;
protected:
// time jitter for aac
SrsTsAacJitter* aac_jitter;
public:
SrsTsCache();
virtual ~SrsTsCache();
public:
/**
* write audio to cache
*/
virtual int cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample);
/**
* write video to muxer.
*/
virtual int cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
private:
virtual int do_cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int do_cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
};
/**
* encode data to ts file.
... ...