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SRS5: MP3: Convert RTMP(MP3) to WebRTC(OPUS). v5.0.118 (#296) (#3340)

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PICK 37867533cd
This commit is contained in:
winlin 2022-12-26 13:24:07 +08:00
parent f82f265ece
commit 6875876349
7 changed files with 98 additions and 25 deletions
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51
trunk/src/app/srs_app_rtc_source.cpp
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@ -717,6 +717,7 @@ SrsRtcFromRtmpBridge::SrsRtcFromRtmpBridge(SrsRtcSource* source)
source_ = source;
format = new SrsRtmpFormat();
codec_ = new SrsAudioTranscoder();
latest_codec_ = SrsAudioCodecIdForbidden;
rtmp_to_rtc = false;
keep_bframe = false;
merge_nalus = false;
@ -766,12 +767,6 @@ srs_error_t SrsRtcFromRtmpBridge::initialize(SrsRequest* r)
// Setup the SPS/PPS parsing strategy.
format->try_annexb_first = _srs_config->try_annexb_first(r->vhost);
int bitrate = 48000; // The output bitrate in bps.
if ((err = codec_->initialize(SrsAudioCodecIdAAC, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate,
bitrate)) != srs_success) {
return srs_error_wrap(err, "init codec");
}
}
keep_bframe = _srs_config->get_rtc_keep_bframe(req->vhost);
@ -831,6 +826,11 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
return srs_error_wrap(err, "format consume audio");
}
// Try to init codec when startup or codec changed.
if (format->acodec && (err = init_codec(format->acodec->id)) != srs_success) {
return srs_error_wrap(err, "init codec");
}
// 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) {
@ -843,14 +843,18 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
return err;
}
// ignore sequence header
srs_assert(format->audio);
if (format->acodec->id == SrsAudioCodecIdMP3) {
return transcode(format->audio);
}
// When drop aac audio packet, never transcode.
if (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.
@ -875,6 +879,35 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
return err;
}
srs_error_t SrsRtcFromRtmpBridge::init_codec(SrsAudioCodecId codec)
{
srs_error_t err = srs_success;
// Ignore if not changed.
if (latest_codec_ == codec) return err;
// Create a new codec.
srs_freep(codec_);
codec_ = new SrsAudioTranscoder();
// Initialize the codec according to the codec in stream.
int bitrate = 48000; // The output bitrate in bps.
if ((err = codec_->initialize(codec, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate, bitrate)) != srs_success) {
return srs_error_wrap(err, "init codec=%d", codec);
}
// Update the latest codec in stream.
if (latest_codec_ == SrsAudioCodecIdForbidden) {
srs_trace("RTMP2RTC: Init audio codec to %d(%s)", codec, srs_audio_codec_id2str(codec).c_str());
} else {
srs_trace("RTMP2RTC: Switch audio codec %d(%s) to %d(%s)", latest_codec_, srs_audio_codec_id2str(latest_codec_).c_str(),
codec, srs_audio_codec_id2str(codec).c_str());
}
latest_codec_ = codec;
return err;
}
srs_error_t SrsRtcFromRtmpBridge::transcode(SrsAudioFrame* audio)
{
srs_error_t err = srs_success;