How to do a live broadcast on the Internet radio
In order to do live broadcast, I do not need much: I have to adjust the sound, so that music will be played and done recording from the microphone simultaneously, also would be great if this will work with Skype. In practice, it is not so simple. In this article, I will tell about settings of my workstation and software that will be used during the broadcast on the Internet radio.
The above setup is a quite suitable for the music broadcasts; furthermore, it was originally made for them.
The overall structure looks as follows: even during the pair broadcasts the ready-made sound is formed on the side of one announcer, then this flow is transmitted to the server, which distributes it to the audience (it is all done by icecast2). Simultaneously, live broadcasts are recorded for the subsequent automated processing.
I use a USB-microphone Samson C01U:
It gives quite acceptable sound, though I think any USB-microphone or headset would be fine. The main problem is that I need to hear a player and what says the other announcer, at that I should not hear myself. The USB-microphones operate with a delay, it confuses and disturbs me. Specifically, this microphone does not have the outlet for monitoring, but even if it was there, it would not change anything: I need to hear the music / co-announcer.
The minimum requirements are:
1) I need to hear everything except myself in the headphones;
2) I need to give the complete flow (including myself) to broadcast.
That is necessary to form two separate audio flows, so it would require two sound cards (hardware or virtual). I just had two sound cards, let us call them snd1 and snd2. Both have 4 outlets front and back, which is important. I form the sound on the one sound card (snd1) that I want to hear myself and from one output (snd1/front) I direct it in the headphones, and from the second one (snd1/back) I sent to the line input of the second card.
Sound transmission from snd1 to snd2 is done by using a short cable. Since a stronger linear signal is transmitted, then the sound quality is not too much affected. On the second card (snd2) is enabled monitoring of linear outlet, and I add to it what I say into the microphone, thus I get ready sound for a transmission. In order to take off a sound from snd2 for recording or digital transmission, the card should be set to a record mode “what you hear”. Previously, this record mode was rare.
Thus, a sound that I send to the server that is complete formed flow, which contains: myself, the second announcer and jingles / interview / music, I take from outlet of card snd2, which is called “what you hear”.
I use on the workstation:
- Foobar2000: music player;
- Skype: connection with the second announcer, guests, and phone calls;
- VST Host: realtime effects for the microphone;
- Audacity: Recording and monitoring of the flow;
- Edcast Standalone: Broadcast of formed flow that is generated by the server.
By the way, if you do not use the hardware sound cards and use a virtual - a set of software, the situation practically will remain unchanged.
Then, a little more details about each of the used programs.
The choice is justified only because a customizable player is familiar to me. During the broadcast it is important to make it easy to start / slow down music and control a volume. In fact, I run two instances of foobar, two players are needed to make smooth transitions between songs or play some jingles for the background of sounding music. I know that there is some specialized DJ software to all this, but I handle two foobars just well.
Here everything is clear except for one thing: the new versions of Skype play only on a front outlet of the sound card, because of this I have to use older versions. At one time, I used the hardware fork, but this method has its disadvantages. In short, it is too quiet and there is not a possibility separately to control the volume of sound that goes to the headphones and snd2. On the other hand, it is very convenient: everybody has it; you can organize conferences, invite guests on the air, and make calls to the mobile phones of the reporters.
This particular model of microphone has a strange approach to the sound: it sounds just to the right channel. In order to correct it there has been added VST host. Additionally, you can simultaneously include, for example, a compressor or some other effects. At the moment, I only fix flaw in my microphone. Another function that performs is audio broadcast from a microphone on snd2. This could be done in some other way, but I use this host. Accordingly, I disable the microphone when it is required (usually, it is in the pauses when the music plays.)
In addition to recording, I still watch for the forming a wave to follow the ratio of the volume levels of myself / music / guests. Recording that I do for myself is just a backup and the main recording is done by the server of radio station. If I process the sound after a broadcast, it is done in Audacity: normalization, compression, and noise removal.
It transmits the generated audio flow to the server, where it is received by icecast2, which is run on our server side. SB Live! – This is the same snd2.
That is all in terms of the workstation. This option is not a dogma, but rather an illustration of how to solve the problem by available resources. Now, I think how to simplify the way using a normal hardware mixer or vice versa transition to the virtual sound cards.
Bonus is a screenshot at the time of broadcasting.
Obviously, you can ask questions, suddenly I forgot something to write about.
“Translated from another resource”
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