I am using BASS_Encode_StartCA with proc set to NULL, this works with 'aac ' why not 'aacf'?
The encoder is then passed to a Caster.

Thanks for your help Ian.

The AAC-HE library is going to cost me 50,000 Euros.
However the company I was looking at buying it from suggested AAC+ELD+SBR as an alternate as "iOS supports encoding to this".

Interestingly enough if I specify 'aach' for the BASS_Encode_StartCAFile call, I get BASS_ERROR_FORMAT.
If I specify 'aacf', I get BASS_ERROR_UNKNOWN.

Any suggestions. Should my source be set for FloatingPoint when using 'aach'?

Thanks Ian.

I am looking into obtaining a static library for HE-AAC encoding and the appropriate licence for using AAC technology in my application.

Hello,

I am using the following code:

BASS_Encode_StartCA(_hMixer, 'adts', 'aac ', 0, 128000, 0, 0);

to start an AAC encoder.

I have a request to implement AAC+ but from what I can see BASS for iOS doesn't support this. Is this correct?

Hello Ian,

The mixer is feeding an encoder that then feeds a shoutcast server.
I know that due to buffers, processing of real-time audio will be slowed.
I'm just curious if there is a way to minimise this delay?

I will get the output of BASS_ChannelGetData later today.

Hello,

I am using BASS_Mixer_ChannelGetLevel to get the volume level of a recording stream connected to a mixer this is then used to display a VU meter.

The VU meter has roughly a 2 second delay, is there anyway to reduce this delay to make it more accurate?

Hello,

I am using the BASS encode plugin to stream to a shoutcast server and I was wondering what the best method would be to get the number of bytes sent? I am wanting to display to the user how much data has been used.



Thanks 

The iPhone 5 uses ARMv7s. Please add this to BASS, BASS_FX, etc.

Thanks Ian

http://streamfinder.com/internet-radio-api/

At the moment I am using a DAMP effect and then a Compressor effect to give loud and quiet songs a "consistent" volume.
Is there any other way to implement AGC (Automatic Gain Control)?


Thank you

Hello Lumoraes,

BASS doesn't have an inbuilt function to get milliseconds elapsed.

If I use a for loop to iterate over all the devices I only get one with a name of "Default" even if I have a USB mic plugged in to the iPad.
Is this because iOS overrides the device "Default" with the USB mic or is it a limitation of Bass for iOS?


Thank you

Hello,

I am broadcasting to a shoutcast server and the volume is so low that I need to turn the volume up quite a bit on the radio.
If I adjust the volume of the mixer up using BASS_BFX_VOLUME then the radio volume stays the same but the sound becomes distorted (clipped).

Any ideas?


Thanks

Maybe Ian could add a BASS_SPEAKER_NONE flag?

Hello,

I have an MP3 file and a microphone input both going to a mixer.
At the moment both the MP3 and mic go to the speaker, I would like only the MP3 to go to the speaker but both to still go to the mixer output.

Basically if you are talking there is a delayed echo of your voice on the speaker and it's distracting.
I would like to remove this echo.

Is there some way to tell BASS to exclude a channel from output?


Thanks

Got it working 

I moved the buffer declaration to outside of the function:

unsigned char mp3_buffer[800000];

Then divide length by 4 as lame needs number of samples

void CALLBACK EncodingDSP(HDSP handle, DWORD channel, void *buffer, DWORD length, void *user)
{
    int write;
    HENCODE encHandle = [(Broadcaster*)user encHandle];
    write = lame_encode_buffer_interleaved(lame, buffer, length/4, mp3_buffer, 800000);
    BASS_Encode_Write(encHandle, mp3_buffer, write);
}

Adding support for this has made me a very happy man
Much thanks Ian.

Ok, here is my latest code:

lame = lame_init();
lame_set_in_samplerate(lame, 44100);
lame_set_mode(lame, JOINT_STEREO);
lame_set_quality(lame, 7);
lame_set_findReplayGain(lame, 0);
lame_set_brate(lame, 128);
lame_set_VBR_min_bitrate_kbps(lame, lame_get_brate(lame));
lame_set_out_samplerate(lame, 44100);
lame_init_params(lame);

BASS_ChannelSetDSP(m_MixerHandle, EncodingDSP, self, -1000);       
encHandle=BASS_Encode_Start(m_MixerHandle, NULL, BASS_ENCODE_PCM|BASS_ENCODE_NOHEAD|BASS_ENCODE_PAUSE, NULL, NULL);

BASS_Encode_CastInit(encHandle, "192.168.0.48:8000",
                                        "ais22b",
                                        BASS_ENCODE_TYPE_MP3,
                                        "TribeFM",
                                        NULL,
                                        "Various",
                                        NULL, 0, 128, NO);

BASS_ChannelPlay(m_MixerHandle, true);

void CALLBACK EncodingDSP(HDSP handle, DWORD channel, void *buffer, DWORD length, void *user)
{
    int write;
    HENCODE encHandle = [(Broadcaster*)user encHandle];
    unsigned char mp3_buffer[8000];
    write = lame_encode_buffer_interleaved(lame, buffer, length/2, mp3_buffer, 8000);
    BASS_Encode_Write(encHandle, mp3_buffer, write);
}

This has the output from BASS back to smooth sounding audio.
However the sound from SHOUTcast is broken up and has crunching noises in it.

I'm assuming it's because of my hacked "No idea what i'm doing" attempt of a callback function.

Please help 

Lame Functions Explained: http://openinnowhere.sourceforge.net/lameonj/javadoc/lame/std/Lame.html

Thanks Ian.

I must be doing things wrong as I get choppy sound from BASS when I do this:

void CALLBACK EncodingDSP(HDSP handle, DWORD channel, void *buffer, DWORD length, void *user)
{
   
    int write;
   
    unsigned char mp3_buffer[length];
    if (length == 0)
        write = lame_encode_flush(lame, mp3_buffer, length);
    else
        write = lame_encode_buffer_interleaved(lame, buffer, length, mp3_buffer, length);

    BASS_Encode_Write((HENCODE)user, mp3_buffer, write); // send data to the encoder/caster
}

Do you think this is due to CPU load or have I done something wrong?


Thanks

Hello,

Is there any way to tell a "Cast" to connect/disconnect and a way to change it's settings (e.g. server, password)?
Or can i simply set it to nil and then re-init a new cast.

Sorry that was a typo.

I was referring to your posted code and how I might go about labeling -30, -20, -15, -10, -7, -5, -3, -2, -1 and 0.
Obviously -30 and 0 are easy, but is there some way I can visually see where the others are?