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