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support config the audio overflow ratio.

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This commit is contained in:
winlin 2015-03-19 17:24:27 +08:00
parent e319da3329
commit 0319e85f99
5 changed files with 43 additions and 16 deletions
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24
trunk/src/app/srs_app_hls.cpp
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@ -168,6 +168,7 @@ SrsHlsMuxer::SrsHlsMuxer()
req = NULL;
handler = NULL;
hls_fragment = hls_window = 0;
hls_aof_ratio = 1.0;
target_duration = 0;
_sequence_no = 0;
current = NULL;
@ -203,7 +204,7 @@ int SrsHlsMuxer::sequence_no()
return _sequence_no;
}
int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path, int fragment, int window)
int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path, int fragment, int window, double aof_ratio)
{
int ret = ERROR_SUCCESS;
@ -213,11 +214,12 @@ int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path,
hls_entry_prefix = entry_prefix;
hls_path = path;
hls_fragment = fragment;
hls_aof_ratio = aof_ratio;
hls_window = window;
// we always keep the target duration increasing.
int max_td = srs_max(target_duration, (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost)));
srs_info("hls update target duration %d=>%d", target_duration, max_td);
srs_info("hls update target duration %d=>%d, aof=%.2f", target_duration, max_td, aof_ratio);
target_duration = max_td;
std::string storage = _srs_config->get_hls_storage(r->vhost);
@ -345,7 +347,7 @@ bool SrsHlsMuxer::is_segment_overflow()
bool SrsHlsMuxer::is_segment_absolutely_overflow()
{
srs_assert(current);
return current->duration >= 2 * hls_fragment;
return current->duration >= hls_aof_ratio * hls_fragment;
}
int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
@ -676,12 +678,14 @@ int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment
std::string entry_prefix = _srs_config->get_hls_entry_prefix(vhost);
// get the hls path config
std::string hls_path = _srs_config->get_hls_path(vhost);
// the audio overflow, for pure audio to reap segment.
double hls_aof_ratio = _srs_config->get_hls_aof_ratio(vhost);
// TODO: FIXME: support load exists m3u8, to continue publish stream.
// for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
// open muxer
if ((ret = muxer->update_config(req, entry_prefix, hls_path, hls_fragment, hls_window)) != ERROR_SUCCESS) {
if ((ret = muxer->update_config(req, entry_prefix, hls_path, hls_fragment, hls_window, hls_aof_ratio)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer update config failed. ret=%d", ret);
return ret;
}
@ -737,17 +741,13 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
}
}
// cache->audio will be free in flush_audio
// so we must check whether it's null ptr.
if (!cache->audio) {
return ret;
}
// TODO: config it.
// in ms, audio delay to flush the audios.
int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
// flush if audio delay exceed
if (pts - cache->audio->start_pts > audio_delay * 90) {
// cache->audio will be free in flush_audio
// so we must check whether it's null ptr.
if (cache->audio && pts - cache->audio->start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
return ret;
}
@ -761,7 +761,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
// @see https://github.com/winlinvip/simple-rtmp-server/issues/151
// we use absolutely overflow of segment to make jwplayer/ffplay happy
// @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
if (muxer->is_segment_absolutely_overflow()) {
if (cache->audio && muxer->is_segment_absolutely_overflow()) {
if ((ret = reap_segment("audio", muxer, cache->audio->pts)) != ERROR_SUCCESS) {
return ret;
}