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RTC: Remove dead code

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This commit is contained in:
winlin 2020-05-13 15:13:14 +08:00
parent c0021ab78a
commit 8efbdec2af
3 changed files with 9 additions and 157 deletions
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135
trunk/src/app/srs_app_rtc.cpp
View file

@ -99,88 +99,6 @@ srs_error_t SrsRtpH264Muxer::filter(SrsSharedPtrMessage* shared_frame, SrsFormat
return err;
}
SrsRtpOpusMuxer::SrsRtpOpusMuxer()
{
codec = NULL;
}
SrsRtpOpusMuxer::~SrsRtpOpusMuxer()
{
srs_freep(codec);
}
srs_error_t SrsRtpOpusMuxer::initialize()
{
srs_error_t err = srs_success;
codec = new SrsAudioRecode(kChannel, kSamplerate);
if (!codec) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "SrsAacOpus init failed");
}
if ((err = codec->initialize()) != srs_success) {
return srs_error_wrap(err, "init codec");
}
return err;
}
// An AAC packet may be transcoded to many OPUS packets.
const int kMaxOpusPackets = 8;
// The max size for each OPUS packet.
const int kMaxOpusPacketSize = 4096;
srs_error_t SrsRtpOpusMuxer::transcode(SrsSharedPtrMessage* shared_audio, char* adts_audio, int nn_adts_audio)
{
srs_error_t err = srs_success;
// Opus packet cache.
static char* opus_payloads[kMaxOpusPackets];
static bool initialized = false;
if (!initialized) {
initialized = true;
static char opus_packets_cache[kMaxOpusPackets][kMaxOpusPacketSize];
opus_payloads[0] = &opus_packets_cache[0][0];
for (int i = 1; i < kMaxOpusPackets; i++) {
opus_payloads[i] = opus_packets_cache[i];
}
}
// Transcode an aac packet to many opus packets.
SrsSample aac;
aac.bytes = adts_audio;
aac.size = nn_adts_audio;
int nn_opus_packets = 0;
int opus_sizes[kMaxOpusPackets];
if ((err = codec->recode(&aac, opus_payloads, opus_sizes, nn_opus_packets)) != srs_success) {
return srs_error_wrap(err, "recode error");
}
// Save OPUS packets in shared message.
if (nn_opus_packets <= 0) {
return err;
}
int nn_max_extra_payload = 0;
SrsSample samples[nn_opus_packets];
for (int i = 0; i < nn_opus_packets; i++) {
SrsSample* p = samples + i;
p->size = opus_sizes[i];
p->bytes = new char[p->size];
memcpy(p->bytes, opus_payloads[i], p->size);
nn_max_extra_payload = srs_max(nn_max_extra_payload, p->size);
}
shared_audio->set_extra_payloads(samples, nn_opus_packets);
shared_audio->set_max_extra_payload(nn_max_extra_payload);
return err;
}
SrsRtc::SrsRtc()
{
req = NULL;
@ -223,12 +141,7 @@ srs_error_t SrsRtc::initialize(SrsRequest* r)
// TODO: FIXME: Support reload and log it.
discard_aac = _srs_config->get_rtc_aac_discard(req->vhost);
rtp_opus_muxer = new SrsRtpOpusMuxer();
if (!rtp_opus_muxer) {
return srs_error_wrap(err, "rtp_opus_muxer nullptr");
}
return rtp_opus_muxer->initialize();
return err;
}
srs_error_t SrsRtc::on_publish()
@ -266,52 +179,6 @@ void SrsRtc::on_unpublish()
enabled = false;
}
srs_error_t SrsRtc::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
{
srs_error_t err = srs_success;
if (!enabled) {
return err;
}
// Ignore if no format->acodec, it means the codec is not parsed, or unknown codec.
// @issue https://github.com/ossrs/srs/issues/1506#issuecomment-562079474
if (!format->acodec) {
return err;
}
// update the hls time, for hls_dispose.
last_update_time = srs_get_system_time();
// ts support audio codec: aac/mp3
SrsAudioCodecId acodec = format->acodec->id;
if (acodec != SrsAudioCodecIdAAC && acodec != SrsAudioCodecIdMP3) {
return err;
}
// When drop aac audio packet, never transcode.
if (discard_aac && acodec == SrsAudioCodecIdAAC) {
return err;
}
// ignore sequence header
srs_assert(format->audio);
char* adts_audio = NULL;
int nn_adts_audio = 0;
// TODO: FIXME: Reserve 7 bytes header when create shared message.
if ((err = aac_raw_append_adts_header(shared_audio, format, &adts_audio, &nn_adts_audio)) != srs_success) {
return srs_error_wrap(err, "aac append header");
}
if (adts_audio) {
err = rtp_opus_muxer->transcode(shared_audio, adts_audio, nn_adts_audio);
srs_freep(adts_audio);
}
return err;
}
srs_error_t SrsRtc::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
{
srs_error_t err = srs_success;

18
trunk/src/app/srs_app_rtc.hpp
View file

@ -45,9 +45,6 @@ const int kRtpPacketSize = 1500;
const uint8_t kOpusPayloadType = 111;
const uint8_t kH264PayloadType = 102;
const int kChannel = 2;
const int kSamplerate = 48000;
// SSRC will rewrite in srs_app_rtc_conn.cpp when send to client.
const uint32_t kAudioSSRC = 1;
const uint32_t kVideoSSRC = 2;
@ -64,19 +61,6 @@ public:
srs_error_t filter(SrsSharedPtrMessage* shared_video, SrsFormat* format);
};
// TODO: FIXME: It's not a muxer, but a transcoder.
class SrsRtpOpusMuxer
{
private:
SrsAudioRecode* codec;
public:
SrsRtpOpusMuxer();
virtual ~SrsRtpOpusMuxer();
virtual srs_error_t initialize();
public:
srs_error_t transcode(SrsSharedPtrMessage* shared_audio, char* adts_audio, int nn_adts_audio);
};
class SrsRtc
{
private:
@ -86,7 +70,6 @@ private:
bool discard_aac;
srs_utime_t last_update_time;
SrsRtpH264Muxer* rtp_h264_muxer;
SrsRtpOpusMuxer* rtp_opus_muxer;
public:
SrsRtc();
virtual ~SrsRtc();
@ -97,7 +80,6 @@ public:
virtual srs_error_t initialize(SrsRequest* r);
virtual srs_error_t on_publish();
virtual void on_unpublish();
virtual srs_error_t on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format);
virtual srs_error_t on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format);
};

13
trunk/src/app/srs_app_rtc_source.cpp
View file

@ -36,6 +36,14 @@
#include <srs_kernel_buffer.hpp>
#include <srs_app_rtc_codec.hpp>
const int kChannel = 2;
const int kSamplerate = 48000;
// An AAC packet may be transcoded to many OPUS packets.
const int kMaxOpusPackets = 8;
// The max size for each OPUS packet.
const int kMaxOpusPacketSize = 4096;
using namespace std;
// TODO: Add this function into SrsRtpMux class.
@ -585,11 +593,6 @@ srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
return source_->on_audio_imp(msg);
}
// An AAC packet may be transcoded to many OPUS packets.
const int kMaxOpusPackets = 8;
// The max size for each OPUS packet.
const int kMaxOpusPacketSize = 4096;
srs_error_t SrsRtcFromRtmpBridger::transcode(SrsSharedPtrMessage* shared_audio, char* adts_audio, int nn_adts_audio)
{
srs_error_t err = srs_success;