RTC: Remove dead code.

pull/1804/head
winlin 5 years ago
parent ede6684f12
commit 16c47056a6

@ -350,18 +350,6 @@ srs_error_t SrsRtcSource::on_video(SrsCommonMessage* shared_video)
{
srs_error_t err = srs_success;
// drop any unknown header video.
// @see https://github.com/ossrs/srs/issues/421
if (!SrsFlvVideo::acceptable(shared_video->payload, shared_video->size)) {
char b0 = 0x00;
if (shared_video->size > 0) {
b0 = shared_video->payload[0];
}
srs_warn("drop unknown header video, size=%d, bytes[0]=%#x", shared_video->size, b0);
return err;
}
// convert shared_video to msg, user should not use shared_video again.
// the payload is transfer to msg, and set to NULL in shared_video.
SrsSharedPtrMessage msg;
@ -377,18 +365,6 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
bool is_aac_sequence_header = SrsFlvAudio::sh(msg->payload, msg->size);
bool is_sequence_header = is_aac_sequence_header;
// whether consumer should drop for the duplicated sequence header.
bool drop_for_reduce = false;
if (is_sequence_header && meta->previous_ash() && _srs_config->get_reduce_sequence_header(req->vhost)) {
if (meta->previous_ash()->size == msg->size) {
drop_for_reduce = srs_bytes_equals(meta->previous_ash()->payload, msg->payload, msg->size);
srs_warn("drop for reduce sh audio, size=%d", msg->size);
}
}
// TODO: FIXME: Support parsing OPUS for RTC.
if ((err = format->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "format consume audio");
@ -403,32 +379,13 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
}
// copy to all consumer
if (!drop_for_reduce) {
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
return srs_error_wrap(err, "consume message");
}
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
return srs_error_wrap(err, "consume message");
}
}
// cache the sequence header of aac, or first packet of mp3.
// for example, the mp3 is used for hls to write the "right" audio codec.
// TODO: FIXME: to refine the stream info system.
if (is_aac_sequence_header || !meta->ash()) {
if ((err = meta->update_ash(msg)) != srs_success) {
return srs_error_wrap(err, "meta consume audio");
}
}
// if atc, update the sequence header to abs time.
if (meta->ash()) {
meta->ash()->timestamp = msg->timestamp;
}
if (meta->data()) {
meta->data()->timestamp = msg->timestamp;
}
return err;
}
@ -456,38 +413,19 @@ srs_error_t SrsRtcSource::on_video_imp(SrsSharedPtrMessage* msg)
rtc->on_unpublish();
}
// whether consumer should drop for the duplicated sequence header.
bool drop_for_reduce = false;
if (is_sequence_header && meta->previous_vsh() && _srs_config->get_reduce_sequence_header(req->vhost)) {
if (meta->previous_vsh()->size == msg->size) {
drop_for_reduce = srs_bytes_equals(meta->previous_vsh()->payload, msg->payload, msg->size);
srs_warn("drop for reduce sh video, size=%d", msg->size);
}
}
// cache the sequence header if h264
if (is_sequence_header && (err = meta->update_vsh(msg)) != srs_success) {
return srs_error_wrap(err, "meta update video");
}
// copy to all consumer
if (!drop_for_reduce) {
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
return srs_error_wrap(err, "consume video");
}
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
return srs_error_wrap(err, "consume video");
}
}
// if atc, update the sequence header to abs time.
if (meta->vsh()) {
meta->vsh()->timestamp = msg->timestamp;
}
if (meta->data()) {
meta->data()->timestamp = msg->timestamp;
}
return err;
}

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