mirror of
https://github.com/YGGverse/xash3d-fwgs.git
synced 2025-01-17 18:40:02 +00:00
engine: client: voice: new version, move to Opus Custom codec
* Despite Opus Custom have strict requirements, it's more barebones, allowing us to use maximum frame size and custom sample rate, without resampling * Encode each frame size to network buffer, allowing smooth voice chat even in 10 FPS * Fix possible buffer overruns, underruns and races with platform side * Revise all usages of offset variables, samples vs bytes
This commit is contained in:
Alibek Omarov
parent
4110ee0928
commit
82ab06efdd
@ -3085,7 +3085,7 @@ void CL_Init( void )
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VID_Init(); // init video
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S_Init(); // init sound
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Voice_Init( "opus", 3 ); // init voice
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Voice_Init( VOICE_DEFAULT_CODEC, 3 ); // init voice
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// unreliable buffer. unsed for unreliable commands and voice stream
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MSG_Init( &cls.datagram, "cls.datagram", cls.datagram_buf, sizeof( cls.datagram_buf ));
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@ -27,7 +27,6 @@ extern poolhandle_t sndpool;
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#define SOUND_22k 22050 // 22khz sample rate
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#define SOUND_32k 32000 // 32khz sample rate
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#define SOUND_44k 44100 // 44khz sample rate
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#define SOUND_48k 48000 // 48khz sample rate
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#define DMA_MSEC_PER_SAMPLE ((float)(1000.0 / SOUND_DMA_SPEED))
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// fixed point stuff for real-time resampling
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@ -15,13 +15,11 @@ GNU General Public License for more details.
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*/
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#include <opus.h>
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#include <opus_custom.h>
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#include "common.h"
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#include "client.h"
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#include "voice.h"
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static wavdata_t *input_file;
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static fs_offset_t input_pos;
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voice_state_t voice = { 0 };
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CVAR_DEFINE_AUTO( voice_enable, "1", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "enable voice chat" );
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@ -31,6 +29,8 @@ CVAR_DEFINE_AUTO( voice_avggain, "0.5", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "automat
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CVAR_DEFINE_AUTO( voice_maxgain, "5.0", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "automatic voice gain control (maximum)" );
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CVAR_DEFINE_AUTO( voice_inputfromfile, "0", FCVAR_PRIVILEGED, "input voice from voice_input.wav" );
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static void Voice_ApplyGainAdjust( int16_t *samples, int count );
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/*
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===============================================================================
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@ -39,25 +39,6 @@ CVAR_DEFINE_AUTO( voice_inputfromfile, "0", FCVAR_PRIVILEGED, "input voice from
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===============================================================================
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*/
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/*
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=========================
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Voice_GetBandwithTypeName
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=========================
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*/
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static const char* Voice_GetBandwidthTypeName( int bandwidthType )
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{
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switch( bandwidthType )
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{
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case OPUS_BANDWIDTH_FULLBAND: return "Full Band (20 kHz)";
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case OPUS_BANDWIDTH_SUPERWIDEBAND: return "Super Wide Band (12 kHz)";
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case OPUS_BANDWIDTH_WIDEBAND: return "Wide Band (8 kHz)";
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case OPUS_BANDWIDTH_MEDIUMBAND: return "Medium Band (6 kHz)";
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case OPUS_BANDWIDTH_NARROWBAND: return "Narrow Band (4 kHz)";
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default: return "Unknown";
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}
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}
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/*
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=========================
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Voice_CodecInfo_f
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@ -68,7 +49,6 @@ static void Voice_CodecInfo_f( void )
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{
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int encoderComplexity;
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opus_int32 encoderBitrate;
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opus_int32 encoderBandwidthType;
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if( !voice.initialized )
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{
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@ -76,72 +56,12 @@ static void Voice_CodecInfo_f( void )
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return;
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}
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opus_encoder_ctl( voice.encoder, OPUS_GET_BITRATE( &encoderBitrate ));
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opus_encoder_ctl( voice.encoder, OPUS_GET_COMPLEXITY( &encoderComplexity ));
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opus_encoder_ctl( voice.encoder, OPUS_GET_BANDWIDTH( &encoderBandwidthType ));
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opus_custom_encoder_ctl( voice.encoder, OPUS_GET_BITRATE( &encoderBitrate ));
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opus_custom_encoder_ctl( voice.encoder, OPUS_GET_COMPLEXITY( &encoderComplexity ));
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Con_Printf( "Encoder:\n" );
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Con_Printf( " Bitrate: %.3f kbps\n", encoderBitrate / 1000.0f );
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Con_Printf( " Complexity: %d\n", encoderComplexity );
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Con_Printf( " Bandwidth: %s\n", Voice_GetBandwidthTypeName( encoderBandwidthType ));
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}
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/*
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=========================
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Voice_GetFrameSize
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=========================
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*/
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static uint Voice_GetFrameSize( float durationMsec )
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{
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return voice.channels * voice.width * (( float )voice.samplerate / ( 1000.0f / durationMsec ));
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}
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/*
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=========================
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Voice_ApplyGainAdjust
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=========================
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*/
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static void Voice_ApplyGainAdjust( opus_int16 *samples, int count )
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{
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float gain, modifiedMax;
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int average, adjustedSample;
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int blockOffset = 0;
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for( ;;)
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{
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int i;
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int localMax = 0;
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int localSum = 0;
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int blockSize = Q_min( count - ( blockOffset + voice.autogain.block_size ), voice.autogain.block_size );
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if( blockSize < 1 )
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break;
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for( i = 0; i < blockSize; ++i )
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{
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int sample = samples[blockOffset + i];
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if( abs( sample ) > localMax ) {
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localMax = abs( sample );
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}
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localSum += sample;
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gain = voice.autogain.current_gain + i * voice.autogain.gain_multiplier;
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adjustedSample = Q_min( 32767, Q_max(( int )( sample * gain ), -32768 ));
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samples[blockOffset + i] = adjustedSample;
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}
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if( blockOffset % voice.autogain.block_size == 0 )
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{
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average = localSum / blockSize;
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modifiedMax = average + ( localMax - average ) * voice_avggain.value;
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voice.autogain.current_gain = voice.autogain.next_gain * voice_scale.value;
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voice.autogain.next_gain = Q_min( 32767.0f / modifiedMax, voice_maxgain.value ) * voice_scale.value;
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voice.autogain.gain_multiplier = ( voice.autogain.next_gain - voice.autogain.current_gain ) / ( voice.autogain.block_size - 1 );
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}
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blockOffset += blockSize;
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}
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}
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/*
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@ -153,11 +73,22 @@ Voice_InitOpusDecoder
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static qboolean Voice_InitOpusDecoder( void )
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{
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int err;
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voice.decoder = opus_decoder_create( voice.samplerate, voice.channels, &err );
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voice.width = sizeof( opus_int16 );
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voice.samplerate = VOICE_OPUS_CUSTOM_SAMPLERATE;
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voice.frame_size = VOICE_OPUS_CUSTOM_FRAME_SIZE;
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voice.custom_mode = opus_custom_mode_create( SOUND_44k, voice.frame_size, &err );
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if( !voice.custom_mode )
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{
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Con_Printf( S_ERROR "Can't create Opus Custom mode: %s\n", opus_strerror( err ));
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return false;
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}
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voice.decoder = opus_custom_decoder_create( voice.custom_mode, VOICE_PCM_CHANNELS, &err );
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if( !voice.decoder )
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{
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Con_Printf( S_ERROR "Can't create Opus encoder: %s", opus_strerror( err ));
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Con_Printf( S_ERROR "Can't create Opus encoder: %s\n", opus_strerror( err ));
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return false;
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}
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@ -173,37 +104,30 @@ Voice_InitOpusEncoder
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static qboolean Voice_InitOpusEncoder( int quality )
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{
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int err;
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int app = quality == 5 ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP;
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voice.encoder = opus_encoder_create( voice.samplerate, voice.channels, app, &err );
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voice.encoder = opus_custom_encoder_create( voice.custom_mode, VOICE_PCM_CHANNELS, &err );
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if( !voice.encoder )
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{
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Con_Printf( S_ERROR "Can't create Opus encoder: %s", opus_strerror( err ));
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Con_Printf( S_ERROR "Can't create Opus encoder: %s\n", opus_strerror( err ));
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return false;
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}
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switch( quality )
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{
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case 1: // 6 kbps, <6 kHz bandwidth
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opus_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 6000 ));
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opus_encoder_ctl( voice.encoder, OPUS_SET_BANDWIDTH( OPUS_BANDWIDTH_MEDIUMBAND ));
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case 1: // 6 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 6000 ));
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break;
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case 2: // 12 kbps, <12 kHz bandwidth
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opus_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 12000 ));
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opus_encoder_ctl( voice.encoder, OPUS_SET_BANDWIDTH( OPUS_BANDWIDTH_SUPERWIDEBAND ));
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case 2: // 12 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 12000 ));
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break;
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case 4: // 64 kbps, full band (20 kHz)
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opus_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 64000 ));
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opus_encoder_ctl( voice.encoder, OPUS_SET_BANDWIDTH( OPUS_BANDWIDTH_FULLBAND ));
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case 4: // 64 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 64000 ));
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break;
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case 5: // 96 kbps, full band (20 kHz)
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opus_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 96000 ));
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opus_encoder_ctl( voice.encoder, OPUS_SET_BANDWIDTH( OPUS_BANDWIDTH_FULLBAND ));
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case 5: // 96 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 96000 ));
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break;
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default: // 36 kbps, <12 kHz bandwidth
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opus_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 36000 ));
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opus_encoder_ctl( voice.encoder, OPUS_SET_BANDWIDTH( OPUS_BANDWIDTH_SUPERWIDEBAND ));
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default: // 36 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 36000 ));
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break;
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}
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@ -220,7 +144,7 @@ static void Voice_ShutdownOpusDecoder( void )
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{
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if( voice.decoder )
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{
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opus_decoder_destroy( voice.decoder );
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opus_custom_decoder_destroy( voice.decoder );
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voice.decoder = NULL;
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}
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}
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@ -235,9 +159,15 @@ static void Voice_ShutdownOpusEncoder( void )
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{
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if( voice.encoder )
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{
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opus_encoder_destroy( voice.encoder );
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opus_custom_encoder_destroy( voice.encoder );
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voice.encoder = NULL;
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}
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if( voice.custom_mode )
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{
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opus_custom_mode_destroy( voice.custom_mode );
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voice.custom_mode = NULL;
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}
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}
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/*
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@ -248,44 +178,72 @@ Voice_GetOpusCompressedData
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*/
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static uint Voice_GetOpusCompressedData( byte *out, uint maxsize, uint *frames )
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{
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uint ofs, size = 0;
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uint ofs = 0, size = 0;
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uint frame_size_bytes = voice.frame_size * voice.width;
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if( input_file )
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if( voice.input_file )
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{
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uint numbytes;
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double updateInterval;
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updateInterval = cl.mtime[0] - cl.mtime[1];
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numbytes = updateInterval * voice.samplerate * voice.width * voice.channels;
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numbytes = Q_min( numbytes, input_file->size - input_pos );
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numbytes = updateInterval * voice.samplerate * voice.width * VOICE_PCM_CHANNELS;
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numbytes = Q_min( numbytes, voice.input_file->size - voice.input_file_pos );
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numbytes = Q_min( numbytes, sizeof( voice.input_buffer ) - voice.input_buffer_pos );
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memcpy( voice.input_buffer + voice.input_buffer_pos, input_file->buffer + input_pos, numbytes );
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memcpy( voice.input_buffer + voice.input_buffer_pos, voice.input_file->buffer + voice.input_file_pos, numbytes );
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voice.input_buffer_pos += numbytes;
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input_pos += numbytes;
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voice.input_file_pos += numbytes;
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}
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for( ofs = 0; voice.input_buffer_pos - ofs >= voice.frame_size && ofs <= voice.input_buffer_pos; ofs += voice.frame_size )
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if( !voice.input_file )
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VoiceCapture_Lock( true );
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for( ofs = 0; voice.input_buffer_pos - ofs >= frame_size_bytes && ofs <= voice.input_buffer_pos; ofs += frame_size_bytes )
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{
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int bytes;
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if( !input_file )
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#if 1
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if( !voice.input_file )
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{
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// adjust gain before encoding, but only for input from voice
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Voice_ApplyGainAdjust((opus_int16*)voice.input_buffer + ofs, voice.frame_size);
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Voice_ApplyGainAdjust((opus_int16*)(voice.input_buffer + ofs), voice.frame_size);
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}
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#endif
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bytes = opus_encode( voice.encoder, (const opus_int16*)(voice.input_buffer + ofs), voice.frame_size / voice.width, out + size, maxsize );
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memmove( voice.input_buffer, voice.input_buffer + voice.frame_size, sizeof( voice.input_buffer ) - voice.frame_size );
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voice.input_buffer_pos -= voice.frame_size;
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bytes = opus_custom_encode( voice.encoder, (const opus_int16 *)( voice.input_buffer + ofs ),
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voice.frame_size, out + size + sizeof( uint16_t ), maxsize );
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if( bytes > 0 )
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{
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size += bytes;
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// write compressed frame size
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*((uint16_t *)&out[size]) = bytes;
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size += bytes + sizeof( uint16_t );
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maxsize -= bytes + sizeof( uint16_t );
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(*frames)++;
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}
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else
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{
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Con_Printf( S_ERROR "%s: failed to encode frame: %s\n", __func__, opus_strerror( bytes ));
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}
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}
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// did we compress anything? update counters
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if( ofs )
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{
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fs_offset_t remaining = voice.input_buffer_pos - ofs;
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// move remaining samples to the beginning of buffer
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memmove( voice.input_buffer, voice.input_buffer + ofs, remaining );
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voice.input_buffer_pos = remaining;
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}
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if( !voice.input_file )
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VoiceCapture_Lock( false );
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return size;
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}
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@ -297,6 +255,53 @@ static uint Voice_GetOpusCompressedData( byte *out, uint maxsize, uint *frames )
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===============================================================================
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*/
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/*
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=========================
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Voice_ApplyGainAdjust
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=========================
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*/
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static void Voice_ApplyGainAdjust( int16_t *samples, int count )
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{
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float gain, modifiedMax;
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int average, adjustedSample, blockOffset = 0;
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for( ;; )
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{
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int i, localMax = 0, localSum = 0;
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int blockSize = Q_min( count - ( blockOffset + voice.autogain.block_size ), voice.autogain.block_size );
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if( blockSize < 1 )
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break;
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for( i = 0; i < blockSize; ++i )
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{
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int sample = samples[blockOffset + i];
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int absSample = abs( sample );
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if( absSample > localMax )
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localMax = absSample;
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localSum += absSample;
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gain = voice.autogain.current_gain + i * voice.autogain.gain_multiplier;
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adjustedSample = Q_min( SHRT_MAX, Q_max(( int )( sample * gain ), SHRT_MIN ));
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samples[blockOffset + i] = adjustedSample;
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}
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if( blockOffset % voice.autogain.block_size == 0 )
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{
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average = localSum / blockSize;
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modifiedMax = average + ( localMax - average ) * voice_avggain.value;
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voice.autogain.current_gain = voice.autogain.next_gain * voice_scale.value;
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voice.autogain.next_gain = Q_min( (float)SHRT_MAX / modifiedMax, voice_maxgain.value ) * voice_scale.value;
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voice.autogain.gain_multiplier = ( voice.autogain.next_gain - voice.autogain.current_gain ) / ( voice.autogain.block_size - 1 );
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}
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blockOffset += blockSize;
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}
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}
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/*
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=========================
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Voice_Status
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@ -367,21 +372,19 @@ Voice_RecordStop
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*/
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void Voice_RecordStop( void )
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{
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if( input_file )
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if( voice.input_file )
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{
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FS_FreeSound( input_file );
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input_file = NULL;
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FS_FreeSound( voice.input_file );
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voice.input_file = NULL;
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}
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VoiceCapture_Activate( false );
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voice.is_recording = false;
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Voice_Status( VOICE_LOCALCLIENT_INDEX, false );
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voice.input_buffer_pos = 0;
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memset( voice.input_buffer, 0, sizeof( voice.input_buffer ));
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if( Voice_IsRecording( ))
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Voice_Status( VOICE_LOCALCLIENT_INDEX, false );
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VoiceCapture_RecordStop();
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voice.is_recording = false;
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}
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/*
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@ -396,25 +399,25 @@ void Voice_RecordStart( void )
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if( voice_inputfromfile.value )
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{
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input_file = FS_LoadSound( "voice_input.wav", NULL, 0 );
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voice.input_file = FS_LoadSound( "voice_input.wav", NULL, 0 );
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if( input_file )
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if( voice.input_file )
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{
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Sound_Process( &input_file, voice.samplerate, voice.width, SOUND_RESAMPLE );
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input_pos = 0;
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Sound_Process( &voice.input_file, voice.samplerate, voice.width, SOUND_RESAMPLE );
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voice.input_file_pos = 0;
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||||
voice.start_time = Sys_DoubleTime();
|
||||
voice.is_recording = true;
|
||||
}
|
||||
else
|
||||
{
|
||||
FS_FreeSound( input_file );
|
||||
input_file = NULL;
|
||||
FS_FreeSound( voice.input_file );
|
||||
voice.input_file = NULL;
|
||||
}
|
||||
}
|
||||
|
||||
if( !Voice_IsRecording() )
|
||||
voice.is_recording = VoiceCapture_RecordStart();
|
||||
voice.is_recording = VoiceCapture_Activate( true );
|
||||
|
||||
if( Voice_IsRecording() )
|
||||
Voice_Status( VOICE_LOCALCLIENT_INDEX, true );
|
||||
@ -460,7 +463,7 @@ Feed the decoded data to engine sound subsystem
|
||||
static void Voice_StartChannel( uint samples, byte *data, int entnum )
|
||||
{
|
||||
SND_ForceInitMouth( entnum );
|
||||
S_RawEntSamples( entnum, samples, voice.samplerate, voice.width, voice.channels, data, 255 );
|
||||
S_RawEntSamples( entnum, samples, voice.samplerate, voice.width, VOICE_PCM_CHANNELS, data, 255 );
|
||||
}
|
||||
|
||||
/*
|
||||
@ -472,12 +475,35 @@ Received encoded voice data, decode it
|
||||
*/
|
||||
void Voice_AddIncomingData( int ent, const byte *data, uint size, uint frames )
|
||||
{
|
||||
int samples;
|
||||
int samples = 0;
|
||||
int ofs = 0;
|
||||
|
||||
if( !voice.decoder )
|
||||
return;
|
||||
|
||||
samples = opus_decode( voice.decoder, data, size, (short *)voice.decompress_buffer, voice.frame_size / voice.width * frames, false );
|
||||
// decode frame by frame
|
||||
for( ;; )
|
||||
{
|
||||
int frame_samples;
|
||||
uint16_t compressed_size;
|
||||
|
||||
// no compressed size mark
|
||||
if( ofs + sizeof( uint16_t ) > size )
|
||||
break;
|
||||
|
||||
compressed_size = *(const uint16_t *)(data + ofs);
|
||||
ofs += sizeof( uint16_t );
|
||||
|
||||
// no frame data
|
||||
if( ofs + compressed_size > size )
|
||||
break;
|
||||
|
||||
frame_samples = opus_custom_decode( voice.decoder, data + ofs, compressed_size,
|
||||
(opus_int16*)voice.decompress_buffer + samples, voice.frame_size );
|
||||
|
||||
ofs += compressed_size;
|
||||
samples += frame_samples;
|
||||
}
|
||||
|
||||
if( samples > 0 )
|
||||
Voice_StartChannel( samples, voice.decompress_buffer, ent );
|
||||
@ -566,7 +592,7 @@ void Voice_Idle( double frametime )
|
||||
{
|
||||
int i;
|
||||
|
||||
if( !voice_enable.value )
|
||||
if( FBitSet( voice_enable.flags, FCVAR_CHANGED ) && !voice_enable.value )
|
||||
{
|
||||
Voice_Shutdown();
|
||||
return;
|
||||
@ -591,19 +617,16 @@ qboolean Voice_Init( const char *pszCodecName, int quality )
|
||||
if( !voice_enable.value )
|
||||
return false;
|
||||
|
||||
Voice_Shutdown();
|
||||
|
||||
if( Q_strcmp( pszCodecName, "opus" ))
|
||||
if( Q_strcmp( pszCodecName, VOICE_OPUS_CUSTOM_CODEC ))
|
||||
{
|
||||
Con_Printf( S_ERROR "Server requested unsupported codec: %s", pszCodecName );
|
||||
Con_Printf( S_ERROR "Server requested unsupported codec: %s\n", pszCodecName );
|
||||
return false;
|
||||
}
|
||||
|
||||
voice.initialized = false;
|
||||
voice.channels = 1;
|
||||
voice.width = 2;
|
||||
voice.samplerate = SOUND_48k;
|
||||
voice.frame_size = Voice_GetFrameSize( 40.0f );
|
||||
// reinitialize only if codec parameters are different
|
||||
if( Q_strcmp( voice.codec, pszCodecName ) && voice.quality != quality )
|
||||
Voice_Shutdown();
|
||||
|
||||
voice.autogain.block_size = 128;
|
||||
|
||||
if( !Voice_InitOpusDecoder( ))
|
||||
@ -617,6 +640,7 @@ qboolean Voice_Init( const char *pszCodecName, int quality )
|
||||
|
||||
// we can hear others players, so it's fine to fail now
|
||||
voice.initialized = true;
|
||||
Q_strncpy( voice.codec, pszCodecName, sizeof( voice.codec ));
|
||||
|
||||
if( !Voice_InitOpusEncoder( quality ))
|
||||
{
|
||||
@ -624,6 +648,8 @@ qboolean Voice_Init( const char *pszCodecName, int quality )
|
||||
return false;
|
||||
}
|
||||
|
||||
voice.quality = quality;
|
||||
|
||||
if( !VoiceCapture_Init( ))
|
||||
Con_Printf( S_WARN "No microphone is available.\n" );
|
||||
|
||||
|
||||
@ -17,15 +17,29 @@ GNU General Public License for more details.
|
||||
#ifndef VOICE_H
|
||||
#define VOICE_H
|
||||
|
||||
#include "common.h"
|
||||
#include "protocol.h" // MAX_CLIENTS
|
||||
#include "sound.h"
|
||||
|
||||
typedef struct OpusDecoder OpusDecoder;
|
||||
typedef struct OpusEncoder OpusEncoder;
|
||||
typedef struct OpusCustomEncoder OpusCustomEncoder;
|
||||
typedef struct OpusCustomDecoder OpusCustomDecoder;
|
||||
typedef struct OpusCustomMode OpusCustomMode;
|
||||
|
||||
#define VOICE_LOOPBACK_INDEX (-2)
|
||||
#define VOICE_LOCALCLIENT_INDEX (-1)
|
||||
|
||||
#define VOICE_PCM_CHANNELS 1 // always mono
|
||||
|
||||
// never change these parameters when using opuscustom
|
||||
#define VOICE_OPUS_CUSTOM_SAMPLERATE SOUND_44k
|
||||
// must follow opus custom requirements
|
||||
// also be divisible with MAX_RAW_SAMPLES
|
||||
#define VOICE_OPUS_CUSTOM_FRAME_SIZE 1024
|
||||
#define VOICE_OPUS_CUSTOM_CODEC "opus_custom_44k_512"
|
||||
|
||||
// a1ba: do not change, we don't have any re-encoding support now
|
||||
#define VOICE_DEFAULT_CODEC VOICE_OPUS_CUSTOM_CODEC
|
||||
|
||||
typedef struct voice_status_s
|
||||
{
|
||||
qboolean talking_ack;
|
||||
@ -34,6 +48,9 @@ typedef struct voice_status_s
|
||||
|
||||
typedef struct voice_state_s
|
||||
{
|
||||
string codec;
|
||||
int quality;
|
||||
|
||||
qboolean initialized;
|
||||
qboolean is_recording;
|
||||
double start_time;
|
||||
@ -42,20 +59,24 @@ typedef struct voice_state_s
|
||||
voice_status_t players_status[MAX_CLIENTS];
|
||||
|
||||
// opus stuff
|
||||
OpusEncoder *encoder;
|
||||
OpusDecoder *decoder;
|
||||
OpusCustomMode *custom_mode;
|
||||
OpusCustomEncoder *encoder;
|
||||
OpusCustomDecoder *decoder;
|
||||
|
||||
// audio info
|
||||
uint channels;
|
||||
uint width;
|
||||
uint samplerate;
|
||||
uint frame_size;
|
||||
uint frame_size; // in samples
|
||||
|
||||
// buffers
|
||||
byte input_buffer[MAX_RAW_SAMPLES];
|
||||
byte output_buffer[MAX_RAW_SAMPLES];
|
||||
byte decompress_buffer[MAX_RAW_SAMPLES];
|
||||
fs_offset_t input_buffer_pos;
|
||||
fs_offset_t input_buffer_pos; // in bytes
|
||||
|
||||
// input from file
|
||||
wavdata_t *input_file;
|
||||
fs_offset_t input_file_pos; // in bytes
|
||||
|
||||
// automatic gain control
|
||||
struct {
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user