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RTC: Refactor source to accept RTP audio packet.

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This commit is contained in:
winlin 2020-05-13 18:43:25 +08:00
parent 377128f4e9
commit 588d17c09d
5 changed files with 44 additions and 30 deletions
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62
trunk/src/app/srs_app_rtc_source.cpp
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@ -34,6 +34,7 @@
#include <srs_protocol_format.hpp>
#include <srs_kernel_buffer.hpp>
#include <srs_app_rtc_codec.hpp>
#include <srs_kernel_rtc_rtp.hpp>
const int kChannel = 2;
const int kSamplerate = 48000;
@ -436,11 +437,25 @@ void SrsRtcSource::set_rtc_publisher(SrsRtcPublisher* v)
rtc_publisher_ = v;
}
srs_error_t SrsRtcSource::on_rtc_audio(SrsSharedPtrMessage* audio)
srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
{
// TODO: FIXME: Merge with on_audio.
// TODO: FIXME: Print key information.
return on_audio_imp(audio);
srs_error_t err = srs_success;
// copy to all consumer
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");
}
}
return err;
}
srs_error_t SrsRtcSource::on_audio2(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
return err;
}
srs_error_t SrsRtcSource::on_video(SrsCommonMessage* shared_video)
@ -477,21 +492,6 @@ srs_error_t SrsRtcSource::on_video(SrsCommonMessage* shared_video)
return on_video_imp(&msg);
}
srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
// copy to all consumer
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");
}
}
return err;
}
srs_error_t SrsRtcSource::on_video_imp(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
@ -634,14 +634,14 @@ srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
}
if (adts_audio) {
err = transcode(msg, adts_audio, nn_adts_audio);
err = transcode(adts_audio, nn_adts_audio);
srs_freep(adts_audio);
}
return source_->on_audio_imp(msg);
return err;
}
srs_error_t SrsRtcFromRtmpBridger::transcode(SrsSharedPtrMessage* msg, char* adts_audio, int nn_adts_audio)
srs_error_t SrsRtcFromRtmpBridger::transcode(char* adts_audio, int nn_adts_audio)
{
srs_error_t err = srs_success;
@ -684,10 +684,22 @@ srs_error_t SrsRtcFromRtmpBridger::transcode(SrsSharedPtrMessage* msg, char* adt
memcpy(p->bytes, opus_payloads[i], p->size);
nn_max_extra_payload = srs_max(nn_max_extra_payload, p->size);
}
msg->set_extra_payloads(samples, nn_opus_packets);
msg->set_max_extra_payload(nn_max_extra_payload);
SrsRtpPacket2* packet = new SrsRtpPacket2();
packet->rtp_header.set_marker(true);
SrsRtpRawPayload* raw = packet->reuse_raw();
raw->payload = new char[p->size];
raw->nn_payload = p->size;
memcpy(raw->payload, opus_payloads[i], p->size);
// When free the RTP packet, should free the bytes allocated here.
packet->original_bytes = raw->payload;
if ((err = source_->on_audio2(packet)) != srs_success) {
return srs_error_wrap(err, "consume opus");
}
}
return err;
}