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dfe385d0 1 parent 77d78eac

for #293, move the simple buffer to kernel.

内嵌 并排对比
正在显示 10 个修改的文件 包含 622 行增加 和 484 行删除
... ... @@ -366,7 +366,7 @@ ModuleLibIncs=(${SRS_OBJS_DIR})
MODULE_FILES=("srs_kernel_error" "srs_kernel_log" "srs_kernel_stream"
"srs_kernel_utility" "srs_kernel_flv" "srs_kernel_codec" "srs_kernel_file"
"srs_kernel_consts" "srs_kernel_aac" "srs_kernel_mp3" "srs_kernel_ts"
"srs_kernel_avc")
"srs_kernel_avc" "srs_kernel_buffer")
KERNEL_INCS="src/kernel"; MODULE_DIR=${KERNEL_INCS} . auto/modules.sh
KERNEL_OBJS="${MODULE_OBJS[@]}"
#
... ...
... ... @@ -22,6 +22,8 @@ file
../../src/kernel/srs_kernel_aac.cpp,
../../src/kernel/srs_kernel_avc.hpp,
../../src/kernel/srs_kernel_avc.cpp,
../../src/kernel/srs_kernel_buffer.hpp,
../../src/kernel/srs_kernel_buffer.cpp,
../../src/kernel/srs_kernel_codec.hpp,
../../src/kernel/srs_kernel_codec.cpp,
../../src/kernel/srs_kernel_consts.hpp,
... ...
... ... @@ -24,25 +24,6 @@ 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
... ... @@ -80,25 +61,6 @@ using namespace std;
// 63000: 700ms, ts_tbn=90000
#define SRS_AUTO_HLS_DELAY 63000
// 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 */
... ... @@ -162,25 +124,6 @@ u_int8_t mpegts_header[] = {
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
... ... @@ -401,69 +344,6 @@ private:
}
};
SrsHlsAacJitter::SrsHlsAacJitter()
{
base_pts = 0;
nb_samples = 0;
// TODO: config it, 0 means no adjust
sync_ms = SRS_CONF_DEFAULT_AAC_SYNC;
}
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();
... ... @@ -963,27 +843,12 @@ int SrsHlsMuxer::create_dir()
SrsHlsCache::SrsHlsCache()
{
aac_jitter = new SrsHlsAacJitter();
ab = new SrsSimpleBuffer();
vb = new SrsSimpleBuffer();
af = new SrsMpegtsFrame();
vf = new SrsMpegtsFrame();
cache = new SrsTsCache();
}
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);
srs_freep(cache);
}
int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment_start_dts)
... ... @@ -1021,7 +886,7 @@ int SrsHlsCache::on_unpublish(SrsHlsMuxer* muxer)
{
int ret = ERROR_SUCCESS;
if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
return ret;
}
... ... @@ -1047,26 +912,17 @@ int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
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();
}
audio_buffer_start_pts = pts;
// write audio to cache.
if ((ret = cache_audio(codec, sample)) != ERROR_SUCCESS) {
if ((ret = cache->cache_audio(codec, pts, 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) {
if (cache->ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
return ret;
}
}
... ... @@ -1075,7 +931,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
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) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
return ret;
}
}
... ... @@ -1087,7 +943,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
// so we reap event when the audio incoming when segment overflow.
// @see https://github.com/winlinvip/simple-rtmp-server/issues/151
if (muxer->is_segment_overflow()) {
if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
if ((ret = reap_segment("audio", muxer, cache->af->pts)) != ERROR_SUCCESS) {
return ret;
}
}
... ... @@ -1095,33 +951,26 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
return ret;
}
int SrsHlsCache::write_video(
SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t dts, SrsCodecSample* sample)
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) {
if ((ret = cache->cache_video(codec, dts, 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) {
if (cache->vf->key && muxer->is_segment_overflow()) {
if ((ret = reap_segment("video", muxer, cache->vf->dts)) != ERROR_SUCCESS) {
return ret;
}
}
// flush video when got one
if ((ret = muxer->flush_video(af, ab, vf, vb)) != ERROR_SUCCESS) {
if ((ret = muxer->flush_video(cache->af, cache->ab, cache->vf, cache->vb)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush video failed. ret=%d", ret);
return ret;
}
... ... @@ -1147,7 +996,7 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
// 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) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
return ret;
}
... ... @@ -1155,185 +1004,6 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
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, 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 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/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;
}
SrsHls::SrsHls(SrsSource* _source)
{
hls_enabled = false;
... ...
... ... @@ -30,19 +30,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#include <srs_core.hpp>
/**
* the public data, event HLS disable, others can use it.
*/
/**
* the flv sample rate map
*/
extern int flv_sample_rates[];
/**
* the aac sample rate map
*/
extern int aac_sample_rates[];
/**
* the HLS section, only available when HLS enabled.
*/
#ifdef SRS_AUTO_HLS
... ... @@ -62,38 +49,8 @@ class SrsPithyPrint;
class SrsSource;
class SrsFileWriter;
class SrsSimpleBuffer;
/**
* 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 SrsHlsAacJitter
{
private:
int64_t base_pts;
int64_t nb_samples;
int sync_ms;
public:
SrsHlsAacJitter();
virtual ~SrsHlsAacJitter();
/**
* 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();
};
class SrsTsAacJitter;
class SrsTsCache;
/**
* write data from frame(header info) and buffer(data) to ts file.
... ... @@ -223,22 +180,15 @@ private:
* 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 SrsHlsAacJitter,
* so we must gather audio frame together, and recalc the timestamp @see SrsTsAacJitter,
* we use a aac jitter to correct the audio pts.
*/
class SrsHlsCache
{
private:
// current frame and buffer
SrsMpegtsFrame* af;
SrsSimpleBuffer* ab;
SrsMpegtsFrame* vf;
SrsSimpleBuffer* vb;
private:
// the audio cache buffer start pts, to flush audio if full.
int64_t audio_buffer_start_pts;
// time jitter for aac
SrsHlsAacJitter* aac_jitter;
SrsTsCache* cache;
public:
SrsHlsCache();
virtual ~SrsHlsCache();
... ... @@ -271,8 +221,6 @@ private:
* so, user must reap_segment then flush_video to hls muxer.
*/
virtual int reap_segment(std::string log_desc, SrsHlsMuxer* muxer, int64_t segment_start_dts);
virtual int cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
};
/**
... ...
... ... @@ -27,6 +27,358 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#include <srs_kernel_log.hpp>
#include <srs_kernel_stream.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_kernel_buffer.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
};
SrsMpegtsFrame::SrsMpegtsFrame()
{
pts = dts = 0;
pid = sid = cc = 0;
key = false;
}
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()
{
... ...
... ... @@ -33,11 +33,30 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#include <srs_kernel_codec.hpp>
class SrsStream;
class SrsAmf0Object;
class SrsMpegtsFrame;
class SrsSimpleBuffer;
class SrsAvcAacCodec;
class SrsCodecSample;
/**
* the public data, event HLS disable, others can use it.
*/
/**
* the flv sample rate map
*/
extern int flv_sample_rates[];
/**
* the aac sample rate map
*/
extern int aac_sample_rates[];
#define __SRS_SRS_MAX_CODEC_SAMPLE 128
#define __SRS_AAC_SAMPLE_RATE_UNSET 15
// in ms, for HLS aac flush the audio
#define SRS_CONF_DEFAULT_AAC_DELAY 100
/**
* the FLV/RTMP supported audio sample size.
* Size of each audio sample. This parameter only pertains to
... ... @@ -70,6 +89,90 @@ 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();
};
/**
* 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.
... ...
/*
The MIT License (MIT)
Copyright (c) 2013-2015 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_kernel_buffer.hpp>
#include <srs_kernel_error.hpp>
#include <srs_kernel_log.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_core_performance.hpp>
SrsSimpleBuffer::SrsSimpleBuffer()
{
}
SrsSimpleBuffer::~SrsSimpleBuffer()
{
}
int SrsSimpleBuffer::length()
{
int len = (int)data.size();
srs_assert(len >= 0);
return len;
}
char* SrsSimpleBuffer::bytes()
{
return (length() == 0)? NULL : &data.at(0);
}
void SrsSimpleBuffer::erase(int size)
{
if (size <= 0) {
return;
}
if (size >= length()) {
data.clear();
return;
}
data.erase(data.begin(), data.begin() + size);
}
void SrsSimpleBuffer::append(const char* bytes, int size)
{
srs_assert(size > 0);
data.insert(data.end(), bytes, bytes + size);
}
... ...
/*
The MIT License (MIT)
Copyright (c) 2013-2015 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.
*/
#ifndef SRS_KERNEL_BUFFER_HPP
#define SRS_KERNEL_BUFFER_HPP
/*
#include <srs_kernel_buffer.hpp>
*/
#include <srs_core.hpp>
#include <vector>
/**
* the simple buffer use vector to append bytes,
* it's for hls and http, and need to be refined in future.
*/
class SrsSimpleBuffer
{
private:
std::vector<char> data;
public:
SrsSimpleBuffer();
virtual ~SrsSimpleBuffer();
public:
/**
* get the length of buffer. empty if zero.
* @remark assert length() is not negative.
*/
virtual int length();
/**
* get the buffer bytes.
* @return the bytes, NULL if empty.
*/
virtual char* bytes();
/**
* erase size of bytes from begin.
* @param size to erase size of bytes.
* clear if size greater than or equals to length()
* @remark ignore size is not positive.
*/
virtual void erase(int size);
/**
* append specified bytes to buffer.
* @param size the size of bytes
* @remark assert size is positive.
*/
virtual void append(const char* bytes, int size);
};
#endif
... ...
... ... @@ -40,47 +40,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
// @see SrsProtocol::read_message_header().
#define SRS_RTMP_MAX_MESSAGE_HEADER 11
SrsSimpleBuffer::SrsSimpleBuffer()
{
}
SrsSimpleBuffer::~SrsSimpleBuffer()
{
}
int SrsSimpleBuffer::length()
{
int len = (int)data.size();
srs_assert(len >= 0);
return len;
}
char* SrsSimpleBuffer::bytes()
{
return (length() == 0)? NULL : &data.at(0);
}
void SrsSimpleBuffer::erase(int size)
{
if (size <= 0) {
return;
}
if (size >= length()) {
data.clear();
return;
}
data.erase(data.begin(), data.begin() + size);
}
void SrsSimpleBuffer::append(const char* bytes, int size)
{
srs_assert(size > 0);
data.insert(data.end(), bytes, bytes + size);
}
#ifdef SRS_PERF_MERGED_READ
IMergeReadHandler::IMergeReadHandler()
{
... ...
... ... @@ -30,47 +30,9 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#include <srs_core.hpp>
#include <vector>
#include <srs_protocol_io.hpp>
#include <srs_core_performance.hpp>
/**
* the simple buffer use vector to append bytes,
* it's for hls and http, and need to be refined in future.
*/
class SrsSimpleBuffer
{
private:
std::vector<char> data;
public:
SrsSimpleBuffer();
virtual ~SrsSimpleBuffer();
public:
/**
* get the length of buffer. empty if zero.
* @remark assert length() is not negative.
*/
virtual int length();
/**
* get the buffer bytes.
* @return the bytes, NULL if empty.
*/
virtual char* bytes();
/**
* erase size of bytes from begin.
* @param size to erase size of bytes.
* clear if size greater than or equals to length()
* @remark ignore size is not positive.
*/
virtual void erase(int size);
/**
* append specified bytes to buffer.
* @param size the size of bytes
* @remark assert size is positive.
*/
virtual void append(const char* bytes, int size);
};
#include <srs_kernel_buffer.hpp>
#ifdef SRS_PERF_MERGED_READ
/**
... ...