RTC: Remove dead code

trunk/src/app/srs_app_rtc.cpp

@@ -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;
@@ -222,13 +140,8 @@ srs_error_t SrsRtc::initialize(SrsRequest* r)
     rtp_h264_muxer->discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
     // 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;

trunk/src/app/srs_app_rtc.hpp

@@ -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);
 };
 

trunk/src/app/srs_app_rtc_source.cpp

@@ -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;