mirror of
https://github.com/DSheirer/sdrtrunk.git
synced 2026-11-12 16:47:31 -05:00
#1170 resolves issue with MP3 encoded audio raspy. Adds auto-clipping of PCM audio prior to encoding and adds 4x new 32-bit input sample rate selections to choose from, in addition to original 4x 16-bit input formats. (#1171)
Co-authored-by: Dennis Sheirer <dsheirer@github.com>
This commit is contained in:
@@ -24,112 +24,107 @@ import javax.sound.sampled.DataLine;
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import javax.sound.sampled.Line;
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import javax.sound.sampled.SourceDataLine;
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/**
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* Audio formats
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*/
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public class AudioFormats
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{
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public static final AudioFormat.Encoding IMBE_ENCODING =
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new AudioFormat.Encoding( "IMBE" );
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public static final boolean LITTLE_ENDIAN = false;
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public static final boolean BIG_ENDIAN = true;
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public static final float IMBE_FRAME_RATE = 50;
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public static final float IMBE_SAMPLE_RATE = 50;
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public static final float PCM_8_KHZ_RATE = 8000;
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public static final float PCM_16_KHZ_RATE = 16000;
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public static final float PCM_22050_HZ_RATE = 22050;
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public static final float PCM_44100_HZ_RATE = 44100;
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public static final float PCM_48_KHZ_RATE = 48000;
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public static final int IMBE_FRAME_SIZE_BYTES = 18;
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public static final int IMBE_SAMPLE_SIZE_BITS = 144;
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public static final int ONE_CHANNEL = 1;
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public static final int TWO_CHANNELS = 2;
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public static final int PCM_SAMPLE_SIZE_BITS = 16;
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public static final int PCM_FRAME_SIZE_BYTES_MONO = 2;
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public static final int PCM_FRAME_SIZE_BYTES_STEREO = 4;
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public static AudioFormat IMBE_AUDIO_FORMAT =
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new AudioFormat( IMBE_ENCODING,
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IMBE_SAMPLE_RATE,
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IMBE_SAMPLE_SIZE_BITS,
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ONE_CHANNEL,
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IMBE_FRAME_SIZE_BYTES,
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IMBE_FRAME_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_8_KHZ_16BITS_MONO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_8_KHZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_MONO,
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PCM_8_KHZ_RATE,
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LITTLE_ENDIAN );
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public static final int PCM_SAMPLE_SIZE_16_BITS = 16;
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public static final int PCM_SAMPLE_SIZE_32_BITS = 32;
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public static final int PCM_FRAME_SIZE_BYTES_16_BIT_MONO = 2;
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public static final int PCM_FRAME_SIZE_BYTES_32_BIT_MONO = 4;
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public static final int PCM_FRAME_SIZE_BYTES_16_BIT_STEREO = 4;
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public static AudioFormat PCM_SIGNED_16_KHZ_16BITS_MONO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_16_KHZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_MONO,
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PCM_16_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_22050_HZ_16BITS_MONO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_22050_HZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_MONO,
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PCM_22050_HZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_44100_HZ_16BITS_MONO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_44100_HZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_MONO,
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PCM_44100_HZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_8KHZ_16BITS_STEREO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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public static AudioFormat PCM_SIGNED_8000_HZ_32_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_8_KHZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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PCM_SAMPLE_SIZE_32_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_32_BIT_MONO,
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PCM_8_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_8000_HZ_16_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_8_KHZ_RATE,
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PCM_SAMPLE_SIZE_16_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_16_BIT_MONO,
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PCM_8_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_16000_HZ_16_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_16_KHZ_RATE,
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PCM_SAMPLE_SIZE_16_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_16_BIT_MONO,
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PCM_16_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_16000_HZ_32_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_16_KHZ_RATE,
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PCM_SAMPLE_SIZE_32_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_32_BIT_MONO,
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PCM_16_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_22050_HZ_16_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_22050_HZ_RATE,
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PCM_SAMPLE_SIZE_16_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_16_BIT_MONO,
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PCM_22050_HZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_22050_HZ_32_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_22050_HZ_RATE,
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PCM_SAMPLE_SIZE_32_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_32_BIT_MONO,
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PCM_22050_HZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_44100_HZ_16_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_44100_HZ_RATE,
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PCM_SAMPLE_SIZE_16_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_16_BIT_MONO,
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PCM_44100_HZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_44100_HZ_32_BIT_MONO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_44100_HZ_RATE,
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PCM_SAMPLE_SIZE_32_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_32_BIT_MONO,
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PCM_44100_HZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_8000_HZ_16BITS_STEREO = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_8_KHZ_RATE,
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PCM_SAMPLE_SIZE_16_BITS,
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TWO_CHANNELS,
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PCM_FRAME_SIZE_BYTES_STEREO,
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PCM_FRAME_SIZE_BYTES_16_BIT_STEREO,
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PCM_8_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_48KHZ_16BITS_MONO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_48_KHZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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ONE_CHANNEL,
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PCM_FRAME_SIZE_BYTES_MONO,
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PCM_48_KHZ_RATE,
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LITTLE_ENDIAN );
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public static AudioFormat PCM_SIGNED_48KHZ_16BITS_STEREO =
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new AudioFormat( AudioFormat.Encoding.PCM_SIGNED,
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PCM_48_KHZ_RATE,
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PCM_SAMPLE_SIZE_BITS,
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TWO_CHANNELS,
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PCM_FRAME_SIZE_BYTES_STEREO,
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PCM_48_KHZ_RATE,
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LITTLE_ENDIAN );
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/**
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* Source Data Line Info for a 48 kHz, 16-bits signed PCM one channel
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*/
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public static final Line.Info MONO_SOURCE_DATALINE_INFO =
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new DataLine.Info( SourceDataLine.class, PCM_SIGNED_8_KHZ_16BITS_MONO);
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new DataLine.Info( SourceDataLine.class, PCM_SIGNED_8000_HZ_16_BIT_MONO);
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/**
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* Source Data Line Info for a 48 kHz, 16-bits signed PCM two channels
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*/
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public static final Line.Info STEREO_SOURCE_DATALINE_INFO =
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new DataLine.Info( SourceDataLine.class, PCM_SIGNED_8KHZ_16BITS_STEREO);
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new DataLine.Info( SourceDataLine.class, PCM_SIGNED_8000_HZ_16BITS_STEREO);
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}
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@@ -18,8 +18,8 @@
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*/
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package io.github.dsheirer.audio.broadcast;
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import io.github.dsheirer.audio.convert.AudioSampleRate;
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import io.github.dsheirer.audio.convert.ISilenceGenerator;
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import io.github.dsheirer.audio.convert.InputAudioFormat;
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import io.github.dsheirer.audio.convert.MP3Setting;
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import io.github.dsheirer.identifier.IdentifierCollection;
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import io.github.dsheirer.util.ThreadPool;
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@@ -73,13 +73,13 @@ public abstract class AudioStreamingBroadcaster<T extends BroadcastConfiguration
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* The last audio packet's metadata is automatically attached to the closed audio recording when it is enqueued for
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* broadcast. That metadata will be updated on the remote server once the audio recording is opened for streaming.
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*/
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public AudioStreamingBroadcaster(T broadcastConfiguration, AudioSampleRate audioSampleRate, MP3Setting mp3Setting)
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public AudioStreamingBroadcaster(T broadcastConfiguration, InputAudioFormat inputAudioFormat, MP3Setting mp3Setting)
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{
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super(broadcastConfiguration);
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mDelay = getBroadcastConfiguration().getDelay();
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mMaximumRecordingAge = getBroadcastConfiguration().getMaximumRecordingAge();
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mSilenceGenerator = BroadcastFactory.getSilenceGenerator(broadcastConfiguration.getBroadcastFormat(),
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audioSampleRate, mp3Setting);
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inputAudioFormat, mp3Setting);
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}
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public void dispose()
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@@ -30,9 +30,9 @@ import io.github.dsheirer.audio.broadcast.shoutcast.v1.ShoutcastV1AudioBroadcast
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import io.github.dsheirer.audio.broadcast.shoutcast.v1.ShoutcastV1Configuration;
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import io.github.dsheirer.audio.broadcast.shoutcast.v2.ShoutcastV2AudioStreamingBroadcaster;
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import io.github.dsheirer.audio.broadcast.shoutcast.v2.ShoutcastV2Configuration;
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import io.github.dsheirer.audio.convert.AudioSampleRate;
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import io.github.dsheirer.audio.convert.IAudioConverter;
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import io.github.dsheirer.audio.convert.ISilenceGenerator;
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import io.github.dsheirer.audio.convert.InputAudioFormat;
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import io.github.dsheirer.audio.convert.MP3AudioConverter;
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import io.github.dsheirer.audio.convert.MP3Setting;
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import io.github.dsheirer.audio.convert.MP3SilenceGenerator;
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@@ -55,29 +55,29 @@ public class BroadcastFactory
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{
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if(configuration != null)
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{
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AudioSampleRate audioSampleRate = userPreferences.getMP3Preference().getAudioSampleRate();
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InputAudioFormat inputAudioFormat = userPreferences.getMP3Preference().getAudioSampleRate();
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MP3Setting mp3Setting = userPreferences.getMP3Preference().getMP3Setting();
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switch(configuration.getBroadcastServerType())
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{
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case BROADCASTIFY_CALL:
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return new BroadcastifyCallBroadcaster((BroadcastifyCallConfiguration)configuration,
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audioSampleRate, mp3Setting, aliasModel);
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inputAudioFormat, mp3Setting, aliasModel);
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case BROADCASTIFY:
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return new IcecastTCPAudioBroadcaster((BroadcastifyFeedConfiguration) configuration,
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audioSampleRate, mp3Setting, aliasModel);
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inputAudioFormat, mp3Setting, aliasModel);
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case ICECAST_TCP:
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return new IcecastTCPAudioBroadcaster((IcecastTCPConfiguration) configuration, audioSampleRate,
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return new IcecastTCPAudioBroadcaster((IcecastTCPConfiguration) configuration, inputAudioFormat,
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mp3Setting, aliasModel);
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case ICECAST_HTTP:
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return new IcecastHTTPAudioBroadcaster((IcecastHTTPConfiguration) configuration, audioSampleRate,
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return new IcecastHTTPAudioBroadcaster((IcecastHTTPConfiguration) configuration, inputAudioFormat,
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mp3Setting, aliasModel);
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case SHOUTCAST_V1:
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return new ShoutcastV1AudioBroadcaster((ShoutcastV1Configuration) configuration, audioSampleRate,
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return new ShoutcastV1AudioBroadcaster((ShoutcastV1Configuration) configuration, inputAudioFormat,
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mp3Setting, aliasModel);
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case SHOUTCAST_V2:
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return new ShoutcastV2AudioStreamingBroadcaster((ShoutcastV2Configuration) configuration,
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audioSampleRate, mp3Setting, aliasModel);
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inputAudioFormat, mp3Setting, aliasModel);
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case UNKNOWN:
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default:
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mLog.info("Unrecognized broadcastAudio configuration: " + configuration.getBroadcastFormat().name());
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@@ -94,13 +94,13 @@ public class BroadcastFactory
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* @param configuration containing the requested output audio format
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* @return audio convert or null
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*/
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public static IAudioConverter getAudioConverter(BroadcastConfiguration configuration, AudioSampleRate audioSampleRate,
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public static IAudioConverter getAudioConverter(BroadcastConfiguration configuration, InputAudioFormat inputAudioFormat,
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MP3Setting mp3Setting)
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{
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switch(configuration.getBroadcastFormat())
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{
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case MP3:
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return new MP3AudioConverter(audioSampleRate, mp3Setting);
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return new MP3AudioConverter(inputAudioFormat, mp3Setting);
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default:
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mLog.info("Unrecognized broadcastAudio format: " + configuration.getBroadcastFormat().name());
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}
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@@ -140,12 +140,12 @@ public class BroadcastFactory
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return null;
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}
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public static ISilenceGenerator getSilenceGenerator(BroadcastFormat format, AudioSampleRate audioSampleRate, MP3Setting mp3Setting)
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public static ISilenceGenerator getSilenceGenerator(BroadcastFormat format, InputAudioFormat inputAudioFormat, MP3Setting mp3Setting)
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{
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switch(format)
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{
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case MP3:
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return new MP3SilenceGenerator(audioSampleRate, mp3Setting);
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return new MP3SilenceGenerator(inputAudioFormat, mp3Setting);
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default:
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throw new IllegalArgumentException("Unrecognized broadcast format [" + format +
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"] can't create silence generator");
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@@ -25,7 +25,7 @@ import io.github.dsheirer.audio.broadcast.AbstractAudioBroadcaster;
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import io.github.dsheirer.audio.broadcast.AudioRecording;
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import io.github.dsheirer.audio.broadcast.BroadcastEvent;
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import io.github.dsheirer.audio.broadcast.BroadcastState;
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import io.github.dsheirer.audio.convert.AudioSampleRate;
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import io.github.dsheirer.audio.convert.InputAudioFormat;
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import io.github.dsheirer.audio.convert.MP3Setting;
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import io.github.dsheirer.gui.playlist.radioreference.RadioReferenceDecoder;
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import io.github.dsheirer.identifier.Form;
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@@ -83,7 +83,7 @@ public class BroadcastifyCallBroadcaster extends AbstractAudioBroadcaster<Broadc
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* @param config to use
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* @param aliasModel for access to aliases
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*/
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public BroadcastifyCallBroadcaster(BroadcastifyCallConfiguration config, AudioSampleRate audioSampleRate,
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public BroadcastifyCallBroadcaster(BroadcastifyCallConfiguration config, InputAudioFormat inputAudioFormat,
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MP3Setting mp3Setting, AliasModel aliasModel)
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{
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super(config);
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@@ -22,7 +22,7 @@ import io.github.dsheirer.alias.AliasModel;
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import io.github.dsheirer.audio.broadcast.AudioStreamingBroadcaster;
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import io.github.dsheirer.audio.broadcast.BroadcastConfiguration;
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import io.github.dsheirer.audio.broadcast.IBroadcastMetadataUpdater;
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import io.github.dsheirer.audio.convert.AudioSampleRate;
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import io.github.dsheirer.audio.convert.InputAudioFormat;
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import io.github.dsheirer.audio.convert.MP3Setting;
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public abstract class IcecastAudioBroadcaster extends AudioStreamingBroadcaster
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@@ -30,10 +30,10 @@ public abstract class IcecastAudioBroadcaster extends AudioStreamingBroadcaster
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private IBroadcastMetadataUpdater mMetadataUpdater;
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private AliasModel mAliasModel;
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public IcecastAudioBroadcaster(BroadcastConfiguration broadcastConfiguration, AudioSampleRate audioSampleRate,
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public IcecastAudioBroadcaster(BroadcastConfiguration broadcastConfiguration, InputAudioFormat inputAudioFormat,
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MP3Setting mp3Setting, AliasModel aliasModel)
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{
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super(broadcastConfiguration, audioSampleRate, mp3Setting);
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super(broadcastConfiguration, inputAudioFormat, mp3Setting);
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mAliasModel = aliasModel;
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}
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@@ -20,7 +20,7 @@ package io.github.dsheirer.audio.broadcast.icecast;
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import io.github.dsheirer.alias.AliasModel;
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import io.github.dsheirer.audio.broadcast.BroadcastState;
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import io.github.dsheirer.audio.convert.AudioSampleRate;
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import io.github.dsheirer.audio.convert.InputAudioFormat;
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import io.github.dsheirer.audio.convert.MP3AudioConverter;
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import io.github.dsheirer.audio.convert.MP3Setting;
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import io.github.dsheirer.properties.SystemProperties;
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@@ -70,10 +70,10 @@ public class IcecastHTTPAudioBroadcaster extends IcecastAudioBroadcaster
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*
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* @param configuration for the Icecast stream
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*/
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public IcecastHTTPAudioBroadcaster(IcecastHTTPConfiguration configuration, AudioSampleRate audioSampleRate,
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public IcecastHTTPAudioBroadcaster(IcecastHTTPConfiguration configuration, InputAudioFormat inputAudioFormat,
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MP3Setting mp3Setting, AliasModel aliasModel)
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{
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super(configuration, audioSampleRate, mp3Setting, aliasModel);
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super(configuration, inputAudioFormat, mp3Setting, aliasModel);
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}
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/**
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@@ -21,7 +21,7 @@ package io.github.dsheirer.audio.broadcast.icecast;
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import io.github.dsheirer.alias.AliasModel;
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import io.github.dsheirer.audio.broadcast.BroadcastState;
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import io.github.dsheirer.audio.broadcast.icecast.codec.IcecastCodecFactory;
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import io.github.dsheirer.audio.convert.AudioSampleRate;
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import io.github.dsheirer.audio.convert.InputAudioFormat;
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import io.github.dsheirer.audio.convert.MP3AudioConverter;
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import io.github.dsheirer.audio.convert.MP3Setting;
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import io.github.dsheirer.properties.SystemProperties;
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@@ -68,10 +68,10 @@ public class IcecastTCPAudioBroadcaster extends IcecastAudioBroadcaster
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*
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* @param configuration for the Icecast stream
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*/
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public IcecastTCPAudioBroadcaster(IcecastTCPConfiguration configuration, AudioSampleRate audioSampleRate,
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public IcecastTCPAudioBroadcaster(IcecastTCPConfiguration configuration, InputAudioFormat inputAudioFormat,
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MP3Setting mp3Setting, AliasModel aliasModel)
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{
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super(configuration, audioSampleRate, mp3Setting, aliasModel);
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super(configuration, inputAudioFormat, mp3Setting, aliasModel);
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}
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/**
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@@ -22,7 +22,7 @@ import io.github.dsheirer.alias.AliasModel;
|
||||
import io.github.dsheirer.audio.broadcast.AudioStreamingBroadcaster;
|
||||
import io.github.dsheirer.audio.broadcast.BroadcastState;
|
||||
import io.github.dsheirer.audio.broadcast.IBroadcastMetadataUpdater;
|
||||
import io.github.dsheirer.audio.convert.AudioSampleRate;
|
||||
import io.github.dsheirer.audio.convert.InputAudioFormat;
|
||||
import io.github.dsheirer.audio.convert.MP3Setting;
|
||||
import io.github.dsheirer.util.ThreadPool;
|
||||
import org.apache.mina.core.RuntimeIoException;
|
||||
@@ -63,10 +63,10 @@ public class ShoutcastV1AudioBroadcaster extends AudioStreamingBroadcaster
|
||||
*
|
||||
* @param configuration for the Shoutcast stream
|
||||
*/
|
||||
public ShoutcastV1AudioBroadcaster(ShoutcastV1Configuration configuration, AudioSampleRate audioSampleRate,
|
||||
public ShoutcastV1AudioBroadcaster(ShoutcastV1Configuration configuration, InputAudioFormat inputAudioFormat,
|
||||
MP3Setting mp3Setting, AliasModel aliasModel)
|
||||
{
|
||||
super(configuration, audioSampleRate, mp3Setting);
|
||||
super(configuration, inputAudioFormat, mp3Setting);
|
||||
mAliasModel = aliasModel;
|
||||
}
|
||||
|
||||
|
||||
@@ -40,7 +40,7 @@ import io.github.dsheirer.audio.broadcast.shoutcast.v2.ultravox.UltravoxMessageF
|
||||
import io.github.dsheirer.audio.broadcast.shoutcast.v2.ultravox.UltravoxMessageType;
|
||||
import io.github.dsheirer.audio.broadcast.shoutcast.v2.ultravox.UltravoxMetadata;
|
||||
import io.github.dsheirer.audio.broadcast.shoutcast.v2.ultravox.UltravoxProtocolFactory;
|
||||
import io.github.dsheirer.audio.convert.AudioSampleRate;
|
||||
import io.github.dsheirer.audio.convert.InputAudioFormat;
|
||||
import io.github.dsheirer.audio.convert.MP3Setting;
|
||||
import io.github.dsheirer.identifier.Form;
|
||||
import io.github.dsheirer.identifier.Identifier;
|
||||
@@ -93,10 +93,10 @@ public class ShoutcastV2AudioStreamingBroadcaster extends AudioStreamingBroadcas
|
||||
*
|
||||
* @param configuration for the Shoutcast V2 stream
|
||||
*/
|
||||
public ShoutcastV2AudioStreamingBroadcaster(ShoutcastV2Configuration configuration, AudioSampleRate audioSampleRate,
|
||||
public ShoutcastV2AudioStreamingBroadcaster(ShoutcastV2Configuration configuration, InputAudioFormat inputAudioFormat,
|
||||
MP3Setting mp3Setting, AliasModel aliasModel)
|
||||
{
|
||||
super(configuration, audioSampleRate, mp3Setting);
|
||||
super(configuration, inputAudioFormat, mp3Setting);
|
||||
mAliasModel = aliasModel;
|
||||
}
|
||||
|
||||
|
||||
@@ -24,14 +24,19 @@ import io.github.dsheirer.audio.AudioFormats;
|
||||
import javax.sound.sampled.AudioFormat;
|
||||
|
||||
/**
|
||||
* Enumeration of (input) audio sample rates for mono audio.
|
||||
* Enumeration of MP3 input audio formats for mono audio.
|
||||
*/
|
||||
public enum AudioSampleRate
|
||||
public enum InputAudioFormat
|
||||
{
|
||||
SR_8000(AudioFormats.PCM_SIGNED_8_KHZ_16BITS_MONO, "8000 Hz (no resample / default)"),
|
||||
SR_16000(AudioFormats.PCM_SIGNED_16_KHZ_16BITS_MONO, "16000 Hz"),
|
||||
SR_22050(AudioFormats.PCM_SIGNED_22050_HZ_16BITS_MONO, "22050 Hz"),
|
||||
SR_44100(AudioFormats.PCM_SIGNED_44100_HZ_16BITS_MONO, "44100 Hz");
|
||||
SR_8000(AudioFormats.PCM_SIGNED_8000_HZ_16_BIT_MONO, "16-Bit 8000 Hz (no resample)"),
|
||||
SR_16000(AudioFormats.PCM_SIGNED_16000_HZ_16_BIT_MONO, "16-Bit 16000 Hz"),
|
||||
SR_22050(AudioFormats.PCM_SIGNED_22050_HZ_16_BIT_MONO, "16-Bit 22050 Hz (default)"),
|
||||
SR_44100(AudioFormats.PCM_SIGNED_44100_HZ_16_BIT_MONO, "16-Bit 44100 Hz"),
|
||||
|
||||
SR_32_8000(AudioFormats.PCM_SIGNED_8000_HZ_32_BIT_MONO, "32-Bit 8000 Hz (no resample)"),
|
||||
SR_32_16000(AudioFormats.PCM_SIGNED_16000_HZ_32_BIT_MONO, "32-Bit 16000 Hz"),
|
||||
SR_32_22050(AudioFormats.PCM_SIGNED_22050_HZ_32_BIT_MONO, "32-Bit 22050 Hz"),
|
||||
SR_32_44100(AudioFormats.PCM_SIGNED_44100_HZ_32_BIT_MONO, "32-Bit 44100 Hz");
|
||||
|
||||
private AudioFormat mAudioFormat;
|
||||
private String mLabel;
|
||||
@@ -40,7 +45,7 @@ public enum AudioSampleRate
|
||||
* Constructs an instance
|
||||
* @param audioFormat for the specified sample rate
|
||||
*/
|
||||
AudioSampleRate(AudioFormat audioFormat, String label)
|
||||
InputAudioFormat(AudioFormat audioFormat, String label)
|
||||
{
|
||||
mAudioFormat = audioFormat;
|
||||
mLabel = label;
|
||||
@@ -49,9 +54,9 @@ public enum AudioSampleRate
|
||||
/**
|
||||
* Default sample rate
|
||||
*/
|
||||
public static AudioSampleRate getDefault()
|
||||
public static InputAudioFormat getDefault()
|
||||
{
|
||||
return SR_8000;
|
||||
return SR_22050;
|
||||
}
|
||||
|
||||
/**
|
||||
@@ -34,7 +34,7 @@ public class LameFactory
|
||||
* @param setting to apply to the encoder
|
||||
* @return lame encoder
|
||||
*/
|
||||
public static LameEncoder getLameEncoder(AudioSampleRate input, MP3Setting setting)
|
||||
public static LameEncoder getLameEncoder(InputAudioFormat input, MP3Setting setting)
|
||||
{
|
||||
return switch(setting)
|
||||
{
|
||||
@@ -52,13 +52,13 @@ public class LameFactory
|
||||
* @param sampleRate to resample to
|
||||
* @return resampler.
|
||||
*/
|
||||
public static RealResampler getResampler(AudioSampleRate sampleRate)
|
||||
public static RealResampler getResampler(InputAudioFormat sampleRate)
|
||||
{
|
||||
return switch(sampleRate)
|
||||
{
|
||||
case SR_16000 -> new RealResampler(8000, 16000, 4096, 512);
|
||||
case SR_22050 -> new RealResampler(8000, 22050, 4096, 512);
|
||||
case SR_44100 -> new RealResampler(8000, 44100, 4096, 512);
|
||||
case SR_16000, SR_32_16000 -> new RealResampler(8000, 16000, 4096, 512);
|
||||
case SR_22050, SR_32_22050 -> new RealResampler(8000, 22050, 4096, 512);
|
||||
case SR_44100, SR_32_44100 -> new RealResampler(8000, 44100, 4096, 512);
|
||||
default -> throw new IllegalArgumentException("Unrecognized sample rate for resampling: " + sampleRate);
|
||||
};
|
||||
}
|
||||
|
||||
@@ -32,6 +32,9 @@ import java.util.Arrays;
|
||||
import java.util.Collections;
|
||||
import java.util.List;
|
||||
|
||||
/**
|
||||
* Converts PCM audio packets to Mono, MP3 encoded audio
|
||||
*/
|
||||
public class MP3AudioConverter implements IAudioConverter
|
||||
{
|
||||
private final static Logger mLog = LoggerFactory.getLogger( MP3AudioConverter.class );
|
||||
@@ -40,25 +43,33 @@ public class MP3AudioConverter implements IAudioConverter
|
||||
private RealResampler mResampler;
|
||||
private ByteArrayOutputStream mMP3EncodedFramesStream = new ByteArrayOutputStream();
|
||||
private byte[] mOutputFramesBuffer;
|
||||
private InputAudioFormat mInputAudioFormat;
|
||||
|
||||
/**
|
||||
* Converts PCM 16-bit Little Endian audio packets to Mono, MP3 compressed audio.
|
||||
* Constructs an instance.
|
||||
*
|
||||
* @param audioSampleRate for the desired input sample rate (resampled from default 8 kHz as needed)
|
||||
* @param inputAudioFormat for the desired input sample rate and bit size (resampled from default 8 kHz as needed)
|
||||
* @param setting to configure the LAME encoder
|
||||
*/
|
||||
public MP3AudioConverter(AudioSampleRate audioSampleRate, MP3Setting setting)
|
||||
public MP3AudioConverter(InputAudioFormat inputAudioFormat, MP3Setting setting)
|
||||
{
|
||||
mEncoder = LameFactory.getLameEncoder(audioSampleRate, setting);
|
||||
mInputAudioFormat = inputAudioFormat;
|
||||
mEncoder = LameFactory.getLameEncoder(inputAudioFormat, setting);
|
||||
|
||||
if(audioSampleRate != AudioSampleRate.SR_8000)
|
||||
//Resampling is only required if desired input sample rate is not system default of 8kHz
|
||||
if(inputAudioFormat != InputAudioFormat.SR_8000 && inputAudioFormat != InputAudioFormat.SR_32_8000)
|
||||
{
|
||||
mResampler = LameFactory.getResampler(audioSampleRate);
|
||||
mResampler = LameFactory.getResampler(inputAudioFormat);
|
||||
}
|
||||
|
||||
mOutputFramesBuffer = new byte[mEncoder.getPCMBufferSize()];
|
||||
}
|
||||
|
||||
/**
|
||||
* Converts the list of PCM audio packets to MP3 encoded.
|
||||
* @param audioPackets of PCM audio sampled at 8 kHz
|
||||
* @return encoded MP3 audio
|
||||
*/
|
||||
public List<byte[]> convert(List<float[]> audioPackets)
|
||||
{
|
||||
List<byte[]> converted = new ArrayList<>();
|
||||
@@ -70,7 +81,17 @@ public class MP3AudioConverter implements IAudioConverter
|
||||
|
||||
for(int x = 0; x < audioPackets.size(); x++)
|
||||
{
|
||||
byte[] bytesToEncode = ConversionUtils.convertToSigned16BitSamples(audioPackets.get(x)).array();
|
||||
byte[] bytesToEncode = null;
|
||||
|
||||
if(mInputAudioFormat.getAudioFormat().getSampleSizeInBits() == 16)
|
||||
{
|
||||
bytesToEncode = ConversionUtils.convertToSigned16BitSamples(audioPackets.get(x)).array();
|
||||
}
|
||||
else
|
||||
{
|
||||
bytesToEncode = ConversionUtils.convertToSigned32BitSamples(audioPackets.get(x)).array();
|
||||
}
|
||||
|
||||
int bytesToEncodePointer = 0;
|
||||
|
||||
int inputChunkSize = FastMath.min(mOutputFramesBuffer.length, bytesToEncode.length);
|
||||
|
||||
@@ -58,7 +58,7 @@ public class MP3FrameInspector
|
||||
public static void main(String[] args)
|
||||
{
|
||||
mLog.info("Starting ...");
|
||||
MP3SilenceGenerator gen = new MP3SilenceGenerator(AudioSampleRate.SR_8000, MP3Setting.getDefault());
|
||||
MP3SilenceGenerator gen = new MP3SilenceGenerator(InputAudioFormat.SR_8000, MP3Setting.getDefault());
|
||||
|
||||
List<byte[]> audio = gen.generate(173);
|
||||
|
||||
|
||||
@@ -31,16 +31,16 @@ public class MP3SilenceGenerator implements ISilenceGenerator
|
||||
public static final boolean CONSTANT_BIT_RATE = false;
|
||||
|
||||
private MP3AudioConverter mMP3AudioConverter;
|
||||
private AudioSampleRate mAudioSampleRate;
|
||||
private InputAudioFormat mInputAudioFormat;
|
||||
private byte[] mPreviousPartialFrameData;
|
||||
|
||||
/**
|
||||
* Generates MP3 audio silence frames
|
||||
*/
|
||||
public MP3SilenceGenerator(AudioSampleRate audioSampleRate, MP3Setting setting)
|
||||
public MP3SilenceGenerator(InputAudioFormat inputAudioFormat, MP3Setting setting)
|
||||
{
|
||||
mAudioSampleRate = audioSampleRate;
|
||||
mMP3AudioConverter = new MP3AudioConverter(audioSampleRate, setting);
|
||||
mInputAudioFormat = inputAudioFormat;
|
||||
mMP3AudioConverter = new MP3AudioConverter(inputAudioFormat, setting);
|
||||
}
|
||||
|
||||
/**
|
||||
@@ -53,7 +53,7 @@ public class MP3SilenceGenerator implements ISilenceGenerator
|
||||
double duration_secs = (double)duration_ms / 1000.0;
|
||||
|
||||
//We generate silence at 8000 kHz, because it gets resampled by the silence generator to the target rate
|
||||
int length = (int)(duration_secs * AudioSampleRate.SR_8000.getSampleRate());
|
||||
int length = (int)(duration_secs * InputAudioFormat.SR_8000.getSampleRate());
|
||||
|
||||
List<float[]> silenceBuffers = new ArrayList<>();
|
||||
|
||||
@@ -94,7 +94,7 @@ public class MP3SilenceGenerator implements ISilenceGenerator
|
||||
{
|
||||
mLog.debug("Starting ...");
|
||||
|
||||
MP3SilenceGenerator generator = new MP3SilenceGenerator(AudioSampleRate.SR_44100, MP3Setting.VBR_7);
|
||||
MP3SilenceGenerator generator = new MP3SilenceGenerator(InputAudioFormat.SR_44100, MP3Setting.VBR_7);
|
||||
|
||||
int x = 100;
|
||||
|
||||
|
||||
@@ -36,7 +36,7 @@ public class MonoAudioOutput extends AudioOutput
|
||||
|
||||
public MonoAudioOutput(Mixer mixer, UserPreferences userPreferences)
|
||||
{
|
||||
super(mixer, MixerChannel.MONO, AudioFormats.PCM_SIGNED_8_KHZ_16BITS_MONO,
|
||||
super(mixer, MixerChannel.MONO, AudioFormats.PCM_SIGNED_8000_HZ_16_BIT_MONO,
|
||||
AudioFormats.MONO_SOURCE_DATALINE_INFO, BUFFER_SIZE, userPreferences);
|
||||
}
|
||||
|
||||
|
||||
@@ -1,6 +1,6 @@
|
||||
/*******************************************************************************
|
||||
* sdrtrunk
|
||||
* Copyright (C) 2014-2017 Dennis Sheirer
|
||||
/*
|
||||
* *****************************************************************************
|
||||
* Copyright (C) 2014-2022 Dennis Sheirer
|
||||
*
|
||||
* This program is free software: you can redistribute it and/or modify
|
||||
* it under the terms of the GNU General Public License as published by
|
||||
@@ -14,8 +14,8 @@
|
||||
*
|
||||
* You should have received a copy of the GNU General Public License
|
||||
* along with this program. If not, see <http://www.gnu.org/licenses/>
|
||||
*
|
||||
******************************************************************************/
|
||||
* ****************************************************************************
|
||||
*/
|
||||
package io.github.dsheirer.audio.playback;
|
||||
|
||||
import io.github.dsheirer.audio.AudioFormats;
|
||||
@@ -36,7 +36,7 @@ public class StereoAudioOutput extends AudioOutput
|
||||
|
||||
public StereoAudioOutput(Mixer mixer, MixerChannel channel, UserPreferences userPreferences)
|
||||
{
|
||||
super(mixer, channel, AudioFormats.PCM_SIGNED_8KHZ_16BITS_STEREO, AudioFormats.STEREO_SOURCE_DATALINE_INFO,
|
||||
super(mixer, channel, AudioFormats.PCM_SIGNED_8000_HZ_16BITS_STEREO, AudioFormats.STEREO_SOURCE_DATALINE_INFO,
|
||||
BUFFER_SIZE, userPreferences);
|
||||
}
|
||||
|
||||
|
||||
Reference in New Issue
Block a user