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RTC: Refine code

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This commit is contained in:
winlin 2021-01-17 20:30:10 +08:00
parent aaa3918a72
commit 09011eea3a
4 changed files with 131 additions and 6 deletions
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85
trunk/src/kernel/srs_kernel_rtc_rtp.cpp
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@ -179,6 +179,58 @@ void SrsRtpExtensionTwcc::set_sn(uint16_t sn)
has_twcc_ = true;
}
SrsRtpExtensionOneByte::SrsRtpExtensionOneByte() : has_ext_(false), id_(0), value_(0)
{
}
void SrsRtpExtensionOneByte::set_id(int id)
{
id_ = id;
has_ext_ = true;
}
void SrsRtpExtensionOneByte::set_value(uint8_t value)
{
value_ = value;
has_ext_ = true;
}
srs_error_t SrsRtpExtensionOneByte::decode(SrsBuffer* buf)
{
srs_error_t err = srs_success;
if (!buf->require(2)) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "requires %d bytes", 2);
}
uint8_t v = buf->read_1bytes();
id_ = (v & 0xF0) >> 4;
uint8_t len = (v & 0x0F);
if(!id_ || len != 0) {
return srs_error_new(ERROR_RTC_RTP, "invalid rtp extension id=%d, len=%d", id_, len);
}
value_ = buf->read_1bytes();
has_ext_ = true;
return err;
}
srs_error_t SrsRtpExtensionOneByte::encode(SrsBuffer* buf)
{
srs_error_t err = srs_success;
if (!buf->require(2)) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "requires %d bytes", 2);
}
uint8_t id_len = (id_ & 0x0F)<< 4 | 0x00;
buf->write_1bytes(id_len);
buf->write_1bytes(value_);
return err;
}
SrsRtpExtensions::SrsRtpExtensions() : has_ext_(false)
{
}
@ -252,6 +304,11 @@ srs_error_t SrsRtpExtensions::decode_0xbede(SrsBuffer* buf)
return srs_error_wrap(err, "decode twcc extension");
}
has_ext_ = true;
} else if (xtype == kRtpExtensionAudioLevel) {
if((err = audio_level_.decode(buf)) != srs_success) {
return srs_error_wrap(err, "decode audio level extension");
}
has_ext_ = true;
} else {
buf->skip(1 + (len + 1));
}
@ -263,6 +320,7 @@ srs_error_t SrsRtpExtensions::decode_0xbede(SrsBuffer* buf)
uint64_t SrsRtpExtensions::nb_bytes()
{
int size = 4 + (twcc_.has_twcc_ext() ? twcc_.nb_bytes() : 0);
size += (audio_level_.exists() ? audio_level_.nb_bytes() : 0);
// add padding
size += (size % 4 == 0) ? 0 : (4 - size % 4);
return size;
@ -281,6 +339,10 @@ srs_error_t SrsRtpExtensions::encode(SrsBuffer* buf)
len += twcc_.nb_bytes();
}
if (audio_level_.exists()) {
len += audio_level_.nb_bytes();
}
int padding_count = (len % 4 == 0) ? 0 : (4 - len % 4);
len += padding_count;
@ -293,6 +355,12 @@ srs_error_t SrsRtpExtensions::encode(SrsBuffer* buf)
}
}
if (audio_level_.exists()) {
if (srs_success != (err = audio_level_.encode(buf))) {
return srs_error_wrap(err, "encode audio level extension");
}
}
// add padding
while(padding_count > 0) {
buf->write_1bytes(0);
@ -331,6 +399,23 @@ srs_error_t SrsRtpExtensions::set_twcc_sequence_number(uint8_t id, uint16_t sn)
return srs_success;
}
srs_error_t SrsRtpExtensions::get_audio_level(uint8_t& level)
{
if(audio_level_.exists()) {
level = audio_level_.get_value();
return srs_success;
}
return srs_error_new(ERROR_RTC_RTP_MUXER, "not find rtp extension audio level");
}
srs_error_t SrsRtpExtensions::set_audio_level(int id, uint8_t level)
{
has_ext_ = true;
audio_level_.set_id(id);
audio_level_.set_value(level);
return srs_success;
}
SrsRtpHeader::SrsRtpHeader()
{
padding_length = 0;