优化G711 rtp打包分片逻辑相关代码

This commit is contained in:
xia-chu
2024-11-29 20:19:30 +08:00
parent f5d5b71731
commit 64285b6b09
3 changed files with 35 additions and 38 deletions

View File

@@ -2,10 +2,10 @@
namespace mediakit {
G711RtpEncoder::G711RtpEncoder(CodecId codec, uint32_t channels){
_cache_frame = FrameImp::create();
_cache_frame->_codec_id = codec;
G711RtpEncoder::G711RtpEncoder(int sample_rate, int channels, int sample_bit) {
_sample_rate = sample_rate;
_channels = channels;
_sample_bit = sample_bit;
}
void G711RtpEncoder::setOpt(int opt, const toolkit::Any &param) {
@@ -24,36 +24,24 @@ void G711RtpEncoder::setOpt(int opt, const toolkit::Any &param) {
}
bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
auto dur = (_cache_frame->size() - _cache_frame->prefixSize()) / (8 * _channels);
auto next_pts = _cache_frame->pts() + dur;
if (next_pts == 0) {
_cache_frame->_pts = frame->pts();
} else {
if ((next_pts + _pkt_dur_ms) < frame->pts()) { // 有丢包超过20ms
_cache_frame->_pts = frame->pts() - dur;
}
}
_cache_frame->_buffer.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
auto ptr = frame->data() + frame->prefixSize();
auto size = frame->size() - frame->prefixSize();
_buffer.append(ptr, size);
_in_size += size;
_in_pts = frame->pts();
auto stamp = _cache_frame->pts();
auto ptr = _cache_frame->data() + _cache_frame->prefixSize();
auto len = _cache_frame->size() - _cache_frame->prefixSize();
auto remain_size = len;
size_t max_size = 160 * _channels * _pkt_dur_ms / 20; // 20 ms per 160 byte
size_t n = 0;
bool mark = false;
while (remain_size >= max_size) {
assert(remain_size >= max_size);
const size_t rtp_size = max_size;
n++;
stamp += _pkt_dur_ms;
RtpCodec::inputRtp(getRtpInfo().makeRtp(TrackAudio, ptr, rtp_size, mark, stamp), false);
ptr += rtp_size;
remain_size -= rtp_size;
if (!_pkt_bytes) {
// G711压缩率固定是2倍
_pkt_bytes = _pkt_dur_ms * _channels * (_sample_bit / 8) * _sample_rate / 1000 / 2;
}
_cache_frame->_buffer.erase(0, n * max_size);
_cache_frame->_pts += (uint64_t)_pkt_dur_ms * n;
return len > 0;
while (_buffer.size() >= _pkt_bytes) {
_out_size += _pkt_bytes;
auto pts = _in_pts - (_in_size - _out_size) * (_pkt_dur_ms / (float)_pkt_bytes);
RtpCodec::inputRtp(getRtpInfo().makeRtp(TrackAudio, _buffer.data(), _pkt_bytes, false, pts), false);
_buffer.erase(0, _pkt_bytes);
}
return true;
}
} // namespace mediakit