mirror of
https://github.com/drasko/open-ameba.git
synced 2024-12-01 18:50:30 +00:00
79 lines
3.3 KiB
C
79 lines
3.3 KiB
C
|
#ifndef _PLAYER_CONFIG_H_
|
||
|
#define _PLAYER_CONFIG_H_
|
||
|
|
||
|
/*
|
||
|
Define the access point name and its password here.
|
||
|
*/
|
||
|
//#define AP_NAME "HOME_AP"
|
||
|
//#define AP_PASS "0123456789"
|
||
|
|
||
|
/* Define stream URL here. For example, the URL to the MP3 stream of a certain Dutch radio station
|
||
|
is http://icecast.omroep.nl/3fm-sb-mp3 . This translates of a server name of "icecast.omroep.nl"
|
||
|
and a path of "/3fm-sb-mp3". The port usually is 80 (the standard HTTP port) */
|
||
|
#if 0
|
||
|
#define PLAY_SERVER "icecast.omroep.nl/3fm-alternative-mp3" // "/3fm-sb-mp3" // "/3fm-serioustalent-mp3" // "/funx-amsterdamfb-bb-mp3" //
|
||
|
#define PLAY_PORT 80
|
||
|
#endif
|
||
|
#if 1
|
||
|
#define PLAY_SERVER "icecast.omroep.nl/3fm-sb-mp3" // "/funx-amsterdamfb-bb-mp3" //
|
||
|
#define PLAY_PORT 80
|
||
|
#endif
|
||
|
#if 0
|
||
|
#define PLAY_SERVER "icecast.omroep.nl/3fm-serioustalent-mp3" // "/funx-amsterdamfb-bb-mp3"
|
||
|
#define PLAY_PORT 80
|
||
|
#endif
|
||
|
/*
|
||
|
Here's a DI.fm stream
|
||
|
*/
|
||
|
#if 0
|
||
|
#define PLAY_SERVER "pub7.di.fm/di_classiceurodance"
|
||
|
#define PLAY_PORT 80
|
||
|
#endif
|
||
|
|
||
|
/* You can use something like this to connect to a local mpd server which has a configured
|
||
|
mp3 output: */
|
||
|
#if 0
|
||
|
#define PLAY_SERVER "192.168.33.128/"
|
||
|
#define PLAY_PORT 8000
|
||
|
#endif
|
||
|
|
||
|
/* You can also play a non-streaming mp3 file that's hosted somewhere. WARNING: If you do this,
|
||
|
make sure to comment out the ADD_DEL_SAMPLES define below, or you'll get too fast a playback
|
||
|
rate! */
|
||
|
#if 0
|
||
|
#define PLAY_SERVER "meuk.spritesserver.nl/Ii.Romanzeandante.mp3"
|
||
|
#define PLAY_PORT 80
|
||
|
#endif
|
||
|
|
||
|
|
||
|
/*Playing a real-time MP3 stream has the added complication of clock differences: if the sample
|
||
|
clock of the server is a bit faster than our sample clock, it will send out mp3 data faster
|
||
|
than we process it and our buffer will fill up. Conversely, if the server clock is slower, we'll
|
||
|
eat up samples quicker than the server provides them and we end up with an empty buffer.
|
||
|
To fix this, the mp3 logic can insert/delete some samples to modify the speed of playback.
|
||
|
If our buffers are filling up too fast (presumably due to a quick sample clock on the other side)
|
||
|
we will increase our playout speed; if our buffers empty too quickly, we will decrease it a bit.
|
||
|
Unfortunately, adding or deleting samples isn't very good for the audio quality. If you
|
||
|
want better quality, turn this off and/or feel free to implement a better algorithm.
|
||
|
WARNING: Don't use this define if you play non-stream files. It will presume the sample clock
|
||
|
on the server side is waaay too fast and will default to playing back the stream too fast.*/
|
||
|
#define ADD_DEL_SAMPLES
|
||
|
|
||
|
|
||
|
/*While connecting an I2S codec to the I2S port of the ESP is obviously the best way to get nice
|
||
|
16-bit sounds out of the ESP, it is possible to run this code without the codec. For
|
||
|
this to work, instead of outputting a 2x16bit PCM sample the DAC can decode, we use the I2S
|
||
|
port as a makeshift 6.5-bit PWM generator. To do this, we map every mp3 sound sample to a
|
||
|
value that has an amount of 1's set that's linearily related to the sound samples value and
|
||
|
then output that value on the I2S port. The net result is that the average analog value on the
|
||
|
I2S data pin corresponds to the value of the MP3 sample we're trying to output. Needless to
|
||
|
say, a hacked 6.5-bit PWM output is going to sound a lot worse than a real I2S codec.*/
|
||
|
#define PWM_HACK96BIT
|
||
|
|
||
|
/*
|
||
|
* Oversamples x2 low ratio stream (>=48k). Only PWM_HACK.
|
||
|
*/
|
||
|
#define OVERSAMPLES
|
||
|
|
||
|
#endif
|