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Code Issues Releases Wiki Activity

for #304, support config default acodec/vcodec. 2.0.118.

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winlin 2015-02-15 18:11:24 +08:00
parent 922150b2cf
commit 78f34ad46f
10 changed files with 114 additions and 381 deletions
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1
README.md
View file

@ -527,6 +527,7 @@ Supported operating systems and hardware:
### SRS 2.0 history
* v2.0, 2015-02-15, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), support config default acodec/vcodec. 2.0.118.
* v2.0, 2015-02-15, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), rewrite hls/ts code, support h.264+mp3 for hls. 2.0.117.
* v2.0, 2015-02-12, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), use stringstream to generate m3u8, add hls_td_ratio. 2.0.116.
* v2.0, 2015-02-11, dev code ZhouGuowen for 2.0.115.

46
trunk/conf/full.conf
View file

@ -495,10 +495,19 @@ vhost with-hls.srs.com {
# the default audio codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for mp3.
# the available audio codec: aac, mp3
# the available audio codec:
# aac, mp3
# default: aac
# TODO: FIXME: update wiki for it.
hls_acodec aac;
# the default video codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for pure audio(without video) to vn.
# the available video codec:
# h264, vn
# default: h264
# TODO: FIXME: update wiki for it.
hls_vcodec h264;
}
}
# the vhost with hls disabled.
@ -713,9 +722,9 @@ vhost example.transcode.srs.com {
filter_complex 'overlay=10:10';
}
# video encoder name. can be:
# libx264: use h.264(libx264) video encoder.
# copy: donot encoder the video stream, copy it.
# vn: disable video output.
# libx264: use h.264(libx264) video encoder.
# copy: donot encoder the video stream, copy it.
# vn: disable video output.
vcodec libx264;
# video bitrate, in kbps
vbitrate 1500;
@ -731,8 +740,8 @@ vhost example.transcode.srs.com {
# high,main,baseline
vprofile main;
# x264 preset, @see x264 -help, can be:
# ultrafast,superfast,veryfast,faster,fast
# medium,slow,slower,veryslow,placebo
# ultrafast,superfast,veryfast,faster,fast
# medium,slow,slower,veryslow,placebo
vpreset medium;
# other x264 or ffmpeg video params
vparams {
@ -745,14 +754,15 @@ vhost example.transcode.srs.com {
refs 10;
}
# audio encoder name. can be:
# libaacplus: use aac(libaacplus) audio encoder.
# copy: donot encoder the audio stream, copy it.
# an: disable audio output.
# libaacplus: use aac(libaacplus) audio encoder.
# libfdk_aac: use aac(libfdk_aac) audio encoder.
# copy: donot encoder the audio stream, copy it.
# an: disable audio output.
acodec libaacplus;
# audio bitrate, in kbps. [16, 72] for libaacplus.
abitrate 70;
# audio sample rate. for flv/rtmp, it must be:
# 44100,22050,11025,5512
# 44100,22050,11025,5512
asample_rate 44100;
# audio channel, 1 for mono, 2 for stereo.
achannels 2;
@ -762,17 +772,17 @@ vhost example.transcode.srs.com {
profile:a aac_low;
}
# output format, can be:
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# other format, for example, mp4/aac whatever.
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# other format, for example, mp4/aac whatever.
# default: flv
oformat flv;
# output stream. variables:
# [vhost] the input stream vhost.
# [port] the intput stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the tanscode engine name.
# [vhost] the input stream vhost.
# [port] the intput stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the tanscode engine name.
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}

19
trunk/src/app/srs_app_config.cpp
View file

@ -1480,7 +1480,7 @@ int SrsConfig::check_config()
for (int j = 0; j < (int)conf->directives.size(); j++) {
string m = conf->at(j)->name.c_str();
if (m != "enabled" && m != "hls_path" && m != "hls_fragment" && m != "hls_window" && m != "hls_on_error"
&& m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio" && m != "hls_acodec"
&& m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio" && m != "hls_acodec" && m != "hls_vcodec"
) {
ret = ERROR_SYSTEM_CONFIG_INVALID;
srs_error("unsupported vhost hls directive %s, ret=%d", m.c_str(), ret);
@ -3349,6 +3349,23 @@ string SrsConfig::get_hls_acodec(string vhost)
return conf->arg0();
}
string SrsConfig::get_hls_vcodec(string vhost)
{
SrsConfDirective* hls = get_hls(vhost);
if (!hls) {
return SRS_CONF_DEFAULT_HLS_VCODEC;
}
SrsConfDirective* conf = hls->get("hls_vcodec");
if (!conf) {
return SRS_CONF_DEFAULT_HLS_VCODEC;
}
return conf->arg0();
}
SrsConfDirective* SrsConfig::get_dvr(string vhost)
{
SrsConfDirective* conf = get_vhost(vhost);

5
trunk/src/app/srs_app_config.hpp
View file

@ -56,6 +56,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#define SRS_CONF_DEFAULT_HLS_STORAGE "disk"
#define SRS_CONF_DEFAULT_HLS_MOUNT "[vhost]/[app]/[stream].m3u8"
#define SRS_CONF_DEFAULT_HLS_ACODEC "aac"
#define SRS_CONF_DEFAULT_HLS_VCODEC "h264"
#define SRS_CONF_DEFAULT_DVR_PATH "./objs/nginx/html"
#define SRS_CONF_DEFAULT_DVR_PLAN_SESSION "session"
#define SRS_CONF_DEFAULT_DVR_PLAN_SEGMENT "segment"
@ -927,6 +928,10 @@ public:
* get the HLS default audio codec.
*/
virtual std::string get_hls_acodec(std::string vhost);
/**
* get the HLS default video codec.
*/
virtual std::string get_hls_vcodec(std::string vhost);
// dvr section
private:
/**

41
trunk/src/app/srs_app_hls.cpp
View file

@ -131,14 +131,14 @@ string SrsHlsCacheWriter::cache()
return data;
}
SrsHlsSegment::SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac)
SrsHlsSegment::SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac, SrsCodecVideo vc)
{
duration = 0;
sequence_no = 0;
segment_start_dts = 0;
is_sequence_header = false;
writer = new SrsHlsCacheWriter(write_cache, write_file);
muxer = new SrsTSMuxer(writer, ac);
muxer = new SrsTSMuxer(writer, ac, vc);
}
SrsHlsSegment::~SrsHlsSegment()
@ -246,19 +246,36 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
// load the default acodec from config.
SrsCodecAudio default_acodec = SrsCodecAudioAAC;
std::string default_acodec_str = _srs_config->get_hls_acodec(req->vhost);
if (default_acodec_str == "mp3") {
default_acodec = SrsCodecAudioMP3;
srs_info("hls: use default mp3 acodec");
} else if (default_acodec_str == "aac") {
default_acodec = SrsCodecAudioAAC;
srs_info("hls: use default aac acodec");
} else {
srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
if (true) {
std::string default_acodec_str = _srs_config->get_hls_acodec(req->vhost);
if (default_acodec_str == "mp3") {
default_acodec = SrsCodecAudioMP3;
srs_info("hls: use default mp3 acodec");
} else if (default_acodec_str == "aac") {
default_acodec = SrsCodecAudioAAC;
srs_info("hls: use default aac acodec");
} else {
srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
}
}
// load the default vcodec from config.
SrsCodecVideo default_vcodec = SrsCodecVideoAVC;
if (true) {
std::string default_vcodec_str = _srs_config->get_hls_vcodec(req->vhost);
if (default_vcodec_str == "h264") {
default_vcodec = SrsCodecVideoAVC;
srs_info("hls: use default h264 vcodec");
} else if (default_vcodec_str == "vn") {
default_vcodec = SrsCodecVideoDisabled;
srs_info("hls: use default vn vcodec for pure audio");
} else {
srs_warn("hls: use h264 for other codec=%s", default_vcodec_str.c_str());
}
}
// new segment.
current = new SrsHlsSegment(should_write_cache, should_write_file, default_acodec);
current = new SrsHlsSegment(should_write_cache, should_write_file, default_acodec, default_vcodec);
current->sequence_no = _sequence_no++;
current->segment_start_dts = segment_start_dts;

2
trunk/src/app/srs_app_hls.hpp
View file

@ -144,7 +144,7 @@ public:
// whether current segement is sequence header.
bool is_sequence_header;
public:
SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac);
SrsHlsSegment(bool write_cache, bool write_file, SrsCodecAudio ac, SrsCodecVideo vc);
virtual ~SrsHlsSegment();
public:
/**

2
trunk/src/core/srs_core.hpp
View file

@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
// current release version
#define VERSION_MAJOR 2
#define VERSION_MINOR 0
#define VERSION_REVISION 117
#define VERSION_REVISION 118
// server info.
#define RTMP_SIG_SRS_KEY "SRS"

5
trunk/src/kernel/srs_kernel_codec.hpp
View file

@ -98,7 +98,10 @@ enum SrsCodecVideo
// set to the zero to reserved, for array map.
SrsCodecVideoReserved = 0,
SrsCodecVideoReserved1 = 1,
SrsCodecVideoReserved2 = 8,
SrsCodecVideoReserved2 = 9,
// for user to disable video, for example, use pure audio hls.
SrsCodecVideoDisabled = 8,
SrsCodecVideoSorensonH263 = 2,
SrsCodecVideoScreenVideo = 3,

356
trunk/src/kernel/srs_kernel_ts.cpp
View file

@ -75,327 +75,6 @@ int aac_sample_rates[] =
7350, 0, 0, 0
};
// @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 */
};
u_int8_t mpegts_header_aac[] = {
0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
/* CRC */
0x2f, 0x44, 0xb9, 0x9b, /* crc for aac */
};
u_int8_t mpegts_header_mp3[] = {
0x03, 0xe1, 0x01, 0xf0, 0x00, /* mp3 */
/* CRC */
0x4e, 0x59, 0x3d, 0x1e, /* crc for mp3 */
};
u_int8_t mpegts_header_padding[] = {
/* 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, SrsCodecAudio acodec)
{
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;
}
if (acodec == SrsCodecAudioAAC) {
if ((ret = writer->write(mpegts_header_aac, sizeof(mpegts_header_aac), NULL)) != ERROR_SUCCESS) {
ret = ERROR_HLS_WRITE_FAILED;
srs_error("write ts file aac header failed. ret=%d", ret);
return ret;
}
} else {
if ((ret = writer->write(mpegts_header_mp3, sizeof(mpegts_header_mp3), NULL)) != ERROR_SUCCESS) {
ret = ERROR_HLS_WRITE_FAILED;
srs_error("write ts file mp3 header failed. ret=%d", ret);
return ret;
}
}
if ((ret = writer->write(mpegts_header_padding, sizeof(mpegts_header_padding), NULL)) != ERROR_SUCCESS) {
ret = ERROR_HLS_WRITE_FAILED;
srs_error("write ts file padding 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->write_pcr) {
p[-1] |= 0x20; // Both Adaption and Payload
*p++ = 7; // size
*p++ = 0x50; // random access + PCR
// @see https://github.com/winlinvip/simple-rtmp-server/issues/311
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_dts_pts(p, flags >> 6, frame->pts);
// dts; // 33bits
if (frame->dts != frame->pts) {
p = write_dts_pts(p, 1, frame->dts);
}
}
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, where dts = frame->dts + delay
// and the pcr should never be negative
// @see https://github.com/winlinvip/simple-rtmp-server/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_dts_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;
write_pcr = false;
}
string srs_ts_stream2string(SrsTsStream stream)
{
switch (stream) {
@ -600,6 +279,7 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
case SrsCodecVideoReserved:
case SrsCodecVideoReserved1:
case SrsCodecVideoReserved2:
case SrsCodecVideoDisabled:
case SrsCodecVideoSorensonH263:
case SrsCodecVideoScreenVideo:
case SrsCodecVideoOn2VP6:
@ -645,9 +325,9 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
// encode the media frame to PES packets over TS.
if (msg->is_audio()) {
return encode_pes(writer, msg, audio_pid, as);
return encode_pes(writer, msg, audio_pid, as, vs == SrsTsStreamReserved);
} else {
return encode_pes(writer, msg, video_pid, vs);
return encode_pes(writer, msg, video_pid, vs, vs == SrsTsStreamReserved);
}
}
@ -711,7 +391,7 @@ int SrsTsContext::encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStrea
return ret;
}
int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t pid, SrsTsStream sid)
int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t pid, SrsTsStream sid, bool pure_audio)
{
int ret = ERROR_SUCCESS;
@ -719,6 +399,11 @@ int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t p
return ret;
}
if (sid != SrsTsStreamVideoH264 && sid != SrsTsStreamAudioMp3 && sid != SrsTsStreamAudioAAC) {
srs_info("ts: ignore the unknown stream, sid=%d", sid);
return ret;
}
SrsTsChannel* channel = get(pid);
srs_assert(channel);
@ -729,9 +414,15 @@ int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t p
while (p < end) {
SrsTsPacket* pkt = NULL;
if (p == start) {
// for pure audio stream, always write pcr.
bool write_pcr = msg->write_pcr;
if (pure_audio && msg->is_audio()) {
write_pcr = true;
}
pkt = SrsTsPacket::create_pes_first(this,
pid, msg->sid, channel->continuity_counter++, msg->discontinuity,
msg->write_pcr? msg->dts:-1, msg->dts, msg->pts, msg->payload->length()
write_pcr? msg->dts:-1, msg->dts, msg->pts, msg->payload->length()
);
} else {
pkt = SrsTsPacket::create_pes_continue(this,
@ -1030,9 +721,13 @@ SrsTsPacket* SrsTsPacket::create_pmt(SrsTsContext* context, int16_t pmt_number,
pmt->current_next_indicator = 1;
pmt->section_number = 0;
pmt->last_section_number = 0;
pmt->PCR_PID = vpid;
pmt->program_info_length = 0;
pmt->infos.push_back(new SrsTsPayloadPMTESInfo(vs, vpid));
if (vs == SrsTsStreamVideoH264) {
pmt->PCR_PID = vpid;
pmt->infos.push_back(new SrsTsPayloadPMTESInfo(vs, vpid));
} else {
pmt->PCR_PID = apid;
}
pmt->infos.push_back(new SrsTsPayloadPMTESInfo(as, apid));
pmt->CRC_32 = 0; // calc in encode.
return pkt;
@ -2916,14 +2611,13 @@ int SrsTsPayloadPMT::psi_encode(SrsStream* stream)
return ret;
}
SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac)
SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac, SrsCodecVideo vc)
{
writer = w;
context = NULL;
acodec = ac;
// default to avc(h.264)
vcodec = SrsCodecVideoAVC;
vcodec = vc;
}
SrsTSMuxer::~SrsTSMuxer()
@ -3295,7 +2989,7 @@ int SrsTsEncoder::initialize(SrsFileWriter* fs)
_fs = fs;
srs_freep(muxer);
muxer = new SrsTSMuxer(fs, SrsCodecAudioAAC);
muxer = new SrsTSMuxer(fs, SrsCodecAudioAAC, SrsCodecVideoAVC);
if ((ret = muxer->open("")) != ERROR_SUCCESS) {
return ret;

18
trunk/src/kernel/srs_kernel_ts.hpp
View file

@ -51,20 +51,6 @@ class SrsTsPacket;
// Transport Stream packets are 188 bytes in length.
#define SRS_TS_PACKET_SIZE 188
// @see: ngx_rtmp_SrsMpegtsFrame_t
class SrsMpegtsFrame
{
public:
int64_t pts;
int64_t dts;
int pid;
int sid;
int cc; // continuity_counter
bool write_pcr;
SrsMpegtsFrame();
};
/**
* the pid of ts packet,
* Table 2-3 - PID table, hls-mpeg-ts-iso13818-1.pdf, page 37
@ -387,7 +373,7 @@ public:
virtual int encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo vc, SrsCodecAudio ac);
private:
virtual int encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as);
virtual int encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t pid, SrsTsStream sid);
virtual int encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t pid, SrsTsStream sid, bool pure_audio);
};
/**
@ -1547,7 +1533,7 @@ private:
SrsFileWriter* writer;
std::string path;
public:
SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac);
SrsTSMuxer(SrsFileWriter* w, SrsCodecAudio ac, SrsCodecVideo vc);
virtual ~SrsTSMuxer();
public:
/**