David Shipman provides this update to the audio mixer add-on
git-svn-id: file:///srv/svn/repos/haiku/trunk/current@808 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
beveloper
File diff suppressed because it is too large
Load Diff
@@ -10,7 +10,7 @@
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// forward declarations
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class input_channel;
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class mixer_input;
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// includes
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@@ -36,15 +36,17 @@ class AudioMixer :
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AudioMixer(BMediaAddOn *addOn);
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~AudioMixer();
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// Our own (AudioMixer) methods
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// AudioMixer support
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bool IsValidDest( media_destination dest);
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bool IsValidSource( media_source source);
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status_t ResolveDest(int32 dest, input_channel *outchannel);
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status_t ResolveDest(media_destination dest, input_channel *outchannel);
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BParameterWeb * BuildParameterWeb(); // used to create the initial 'master' web
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void MakeWebForInput(char *name, media_format format);
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void AllocateBuffers();
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status_t FillMixBuffer(void *outbuffer, size_t size);
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// BMediaNode methods
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@@ -185,9 +187,15 @@ class AudioMixer :
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private:
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thread_id fThread; // for launching mixthread
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media_input fInput; // descriptor of single input (temp)
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media_output fOutput; // single output (temp)
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float * fMasterGainScale; // array used for scaling output - _not_ for display!
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float * fMasterGainDisplay; // array of volume values for display purposes
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bigtime_t fMasterGainDisplayLastChange;
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int fMasterMuteValue;
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bigtime_t fMasterMuteValueLastChange;
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BMediaAddOn * fAddOn;
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BParameterWeb * fWeb; // local pointer to parameterweb
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@@ -200,8 +208,9 @@ class AudioMixer :
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media_format fPrefInputFormat, fPrefOutputFormat;
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BBufferGroup * fBufferGroup;
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BList fMixerInputs;
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BList fInputChannels;
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int fNextFreeID;
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};
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@@ -0,0 +1,452 @@
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#include "AudioMixer.h"
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#include "IOStructures.h"
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#include <media/RealtimeAlloc.h>
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#include <media/Buffer.h>
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#include <media/TimeSource.h>
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#include <media/ParameterWeb.h>
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#include <string.h>
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#include <stdio.h>
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#include <stdlib.h>
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status_t
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AudioMixer::FillMixBuffer(void *outbuffer, size_t ioSize)
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{
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int32 outChannels = fOutput.format.u.raw_audio.channel_count;
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// we have an output buffer - now it needs to be filled!
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switch (fOutput.format.u.raw_audio.format)
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{
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case media_raw_audio_format::B_AUDIO_FLOAT:
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{
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float *outdata = (float*)outbuffer;
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memset(outdata, 0, ioSize); // CHANGE_THIS
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int sampleCount = int(fOutput.format.u.raw_audio.buffer_size / sizeof(float));
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for (int s = 0; s < sampleCount; s++)
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{
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outdata[s] = 0.0; // CHANGE_THIS
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}
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int mixerInputCount = fMixerInputs.CountItems();
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for (int c = 0; c < mixerInputCount; c++)
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{
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mixer_input *channel = (mixer_input *)fMixerInputs.ItemAt(c);
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if (channel->enabled == true) // still broken... FIX_THIS
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{
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int32 inChannels = channel->fInput.format.u.raw_audio.channel_count;
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bool split = outChannels > inChannels;
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bool mix = outChannels < inChannels;
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if (fOutput.format.u.raw_audio.frame_rate == channel->fInput.format.u.raw_audio.frame_rate)
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{
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switch(channel->fInput.format.u.raw_audio.format)
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{
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case media_raw_audio_format::B_AUDIO_FLOAT:
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{
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float *indata = (float *)channel->fData;
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if (split)
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{ } // ADD_THIS
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else if (mix)
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{ } // ADD_THIS
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else
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{
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int baseOffset = channel->fEventOffset / 4;
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int maxOffset = int(channel->fDataSize / 4);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int s = 0; s < sampleCount; s++)
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{
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outdata[s] = indata[inputSample];
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indata[inputSample] = 0;
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if (s == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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channel->fEventOffset = (channel->fEventOffset + fOutput.format.u.raw_audio.buffer_size) %
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channel->fDataSize;
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}
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break;
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}
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case media_raw_audio_format::B_AUDIO_SHORT:
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{
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int16 *indata = (int16 *)channel->fData;
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if (split)
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{ }
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else if (mix)
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{ }
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else
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{
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int baseOffset = channel->fEventOffset / 2;
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int maxOffset = int(channel->fDataSize / 2);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int s = 0; s < sampleCount; s++)
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{
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outdata[s] = outdata[s] + (indata[inputSample] / 32768.0); // CHANGE_THIS indata[inputSample] = 0;
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if (s == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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channel->fEventOffset = (channel->fEventOffset + fOutput.format.u.raw_audio.buffer_size / 2) %
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channel->fDataSize;
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}
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break;
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}
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} // input format
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} // samplerate
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} // data available
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} // channel loop
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/*
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// The buffer is done - we still need to scale for the MainVolume...
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// then check to see if anything is clipping, adjust if needed
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*/
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for (int frameStart = 0; frameStart < sampleCount; frameStart += outChannels)
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{
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for (int channel = 0; channel < outChannels; channel ++)
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{
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int sample = frameStart + channel;
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outdata[sample] = outdata[sample] * fMasterGainScale[channel];
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//if (outdata[sample] > 1.0)
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// outdata[sample] = 1.0;
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// else if (outdata[sample] < -1.0)
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// outdata[sample] = -1.0;
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}
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}
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break;
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} // cur-case
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case media_raw_audio_format::B_AUDIO_SHORT:
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{
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int16 *outdata = (int16*)outbuffer;
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int sampleCount = int(fOutput.format.u.raw_audio.buffer_size / sizeof(int16));
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int *clipoffset = new int[sampleCount]; // keep a running tally of +/- clipping so we don't get rollaround distortion
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// we only need this for int/char audio types - not float
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memset(outdata, 0, ioSize);
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memset(clipoffset, 0, sizeof(int) * sampleCount);
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// for (int s = 0; s < sampleCount; s++)
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// {
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// clipoffset[s] = 0;
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// }
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int mixerInputs = fMixerInputs.CountItems();
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for (int c = 0; c < mixerInputs; c++)
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{
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mixer_input *channel = (mixer_input *)fMixerInputs.ItemAt(c);
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if (channel->enabled == true) // only use if there are buffers waiting (seems to be broken atm...)
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{
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// if (channel->fProducerDataStatus == B_DATA_AVAILABLE)
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// printf("B_DATA_AVAILABLE\n");
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// else if (channel->fProducerDataStatus == B_DATA_NOT_AVAILABLE)
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// printf("B_DATA_NOT_AVAILABLE\n");
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// else if (channel->fProducerDataStatus == B_PRODUCER_STOPPED)
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// printf("B_PRODUCER_STOPPED\n");
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int32 inChannels = channel->fInput.format.u.raw_audio.channel_count;
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bool split = outChannels > inChannels;
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bool mix = outChannels < inChannels;
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if (fOutput.format.u.raw_audio.frame_rate == channel->fInput.format.u.raw_audio.frame_rate)
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{
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switch(channel->fInput.format.u.raw_audio.format)
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{
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case media_raw_audio_format::B_AUDIO_FLOAT:
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{
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float *indata = (float *)channel->fData;
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if (split)
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{
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int baseOffset = channel->fEventOffset / 4;
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int maxOffset = int(channel->fDataSize / 4);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int s = 0; s < sampleCount; s += 2)
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{
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if ((outdata[s] + int16(32767 * indata[inputSample])) > 32767)
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clipoffset[s] = clipoffset[s] + 1;
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else if ((outdata[s] + int16(32767 * indata[inputSample])) < -32768)
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clipoffset[s] = clipoffset[s] - 1;
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if ((outdata[s + 1] + int16(32767 * indata[inputSample])) > 32767)
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clipoffset[s + 1] = clipoffset[s + 1] + 1;
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else if ((outdata[s] + int16(32767 * indata[inputSample])) < -32768)
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clipoffset[s + 1] = clipoffset[s + 1] - 1;
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outdata[s] = outdata[s] + int16(32767 * indata[inputSample]); // CHANGE_THIS mixing
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outdata[s + 1] = outdata[s];
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indata[inputSample] = 0;
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if (s == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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}
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else if (mix)
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{}
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else
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{
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int baseOffset = channel->fEventOffset / 4;
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int maxOffset = int(channel->fDataSize / 4);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int frameStart = 0; frameStart < sampleCount; frameStart += outChannels)
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{
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for (int chan = 0; chan < outChannels; chan ++)
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{
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int sample = frameStart + chan;
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int inputValue = (32767 * indata[inputSample] * channel->fGainScale[chan]) + outdata[sample];
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if (inputValue > 32767)
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clipoffset[sample] = clipoffset[sample] + 1;
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else if (inputValue < -32768)
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clipoffset[sample] = clipoffset[sample] - 1;
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outdata[sample] = inputValue; // CHANGE_THIS
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indata[inputSample] = 0;
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if (sample == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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}
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channel->fEventOffset += (fOutput.format.u.raw_audio.buffer_size * 2);
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if (channel->fEventOffset >= channel->fDataSize)
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channel->fEventOffset -= channel->fDataSize;
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}
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break;
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}
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case media_raw_audio_format::B_AUDIO_SHORT:
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{
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int16 *indata = (int16 *)channel->fData;
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if (split)
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{ }
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else if (mix)
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{ }
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else
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{
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int baseOffset = channel->fEventOffset / 2;
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int maxOffset = int(channel->fDataSize / 2);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int frameStart = 0; frameStart < sampleCount; frameStart += outChannels)
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{
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for (int chan = 0; chan < outChannels; chan ++)
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{
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int sample = frameStart + chan;
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int clipTest = (outdata[sample] + (indata[inputSample] * channel->fGainScale[chan]));
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if (clipTest > 32767)
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clipoffset[sample] = clipoffset[sample] + 1;
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else if (clipTest < -32768)
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clipoffset[sample] = clipoffset[sample] - 1;
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outdata[sample] = int16(outdata[sample] + (indata[inputSample] * channel->fGainScale[chan]));
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indata[inputSample] = 0;
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if (sample == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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}
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channel->fEventOffset = (channel->fEventOffset + fOutput.format.u.raw_audio.buffer_size);
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if (channel->fEventOffset >= channel->fDataSize)
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channel->fEventOffset -= channel->fDataSize;
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}
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break;
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}
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case media_raw_audio_format::B_AUDIO_INT:
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{
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int32 *indata = (int32 *)channel->fData;
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if (split)
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{
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int baseOffset = channel->fEventOffset / 4;
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int maxOffset = int(channel->fDataSize / 4);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int s = 0; s < sampleCount; s += 2)
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{
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if ((outdata[s] + (indata[inputSample] / 65536)) > 32767)
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clipoffset[s] = clipoffset[s] + 1;
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else if ((outdata[s] + (indata[inputSample] / 65536)) < -32768)
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clipoffset[s] = clipoffset[s] - 1;
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if ((outdata[s + 1] + (indata[inputSample] / 65536)) > 32767)
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clipoffset[s + 1] = clipoffset[s + 1] + 1;
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else if ((outdata[s + 1] + (indata[inputSample] / 65536)) < -32768)
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clipoffset[s + 1] = clipoffset[s + 1] - 1;
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outdata[s] = int16(outdata[s] + (indata[inputSample] / 65535)); // CHANGE_THIS mixing
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outdata[s + 1] = outdata[s];
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indata[inputSample] = 0;
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if (s == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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}
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else if (mix)
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{ }
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else
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{
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int baseOffset = channel->fEventOffset / 4;
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int maxOffset = int(channel->fDataSize / 4);
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int offsetWrap = maxOffset - baseOffset;
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int inputSample = baseOffset;
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for (int s = 0; s < sampleCount; s++)
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{
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if ((outdata[s] + (indata[inputSample] / 65536)) > 32767)
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clipoffset[s] = clipoffset[s] + 1;
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else if ((outdata[s] + (indata[inputSample] / 65536)) < -32768)
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clipoffset[s] = clipoffset[s] - 1;
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outdata[s] = int16(outdata[s] + (indata[inputSample] / 65536)); // CHANGE_THIS mixing
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indata[inputSample] = 0;
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if (s == offsetWrap)
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inputSample = 0;
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else
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inputSample ++;
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}
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}
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channel->fEventOffset = (channel->fEventOffset + (fOutput.format.u.raw_audio.buffer_size * 2)) %
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channel->fDataSize;
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break;
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}
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}
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}
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}
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}
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// use our clipoffset to determine correct limits
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for (int frameStart = 0; frameStart < sampleCount; frameStart += outChannels)
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{
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for (int channel = 0; channel < outChannels; channel ++)
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{
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int sample = frameStart + channel;
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int scaledSample = (outdata[sample] + (65536 * clipoffset[sample])) * fMasterGainScale[channel];
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if (scaledSample < -32768)
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outdata[sample] = -32768;
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else if (scaledSample > 32767)
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outdata[sample] = 32767;
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else
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outdata[sample] = scaledSample; //(int16)
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}
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}
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delete clipoffset;
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}
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}
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return B_OK;
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}
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@@ -9,19 +9,35 @@
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#ifndef _IO_STRUCT_H
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#define _IO_STRUCT_H
|
||||
|
||||
struct input_channel
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||||
class mixer_input
|
||||
{
|
||||
|
||||
public:
|
||||
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mixer_input(media_input &input);
|
||||
~mixer_input();
|
||||
|
||||
media_input fInput;
|
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int32 fProducerDataStatus; // status of upstream data
|
||||
bool DataAvailable; // there is a buffer ready for mixing
|
||||
bool enabled; // there is a buffer ready for mixing
|
||||
char * fData;
|
||||
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size_t fEventOffset; // offset (in bytes)
|
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|
||||
float * fGainScale; // multiplier for gain - computed at paramchange
|
||||
float * fGainDisplay; // the value that will be displayed for gain
|
||||
bigtime_t fGainDisplayLastChange;
|
||||
|
||||
|
||||
int fMuteValue; // the value of the 'mute' control
|
||||
bigtime_t fMuteValueLastChange;
|
||||
|
||||
float fPanValue; // value of 'pan' control (only if mono)
|
||||
bigtime_t fPanValueLastChange;
|
||||
|
||||
|
||||
size_t fEventOffset; // offset (in bytes) of the start of the next
|
||||
// _output_ buffer - use this for keeping in sync
|
||||
size_t fDataSize; // size of ringbuffer
|
||||
|
||||
};
|
||||
|
||||
#endif;
|
||||
#endif;
|
||||
@@ -5,6 +5,7 @@ UsePrivateHeaders media ;
|
||||
Addon mixer.media_addon : media :
|
||||
AudioMixer.cpp
|
||||
MixerAddOn.cpp
|
||||
FillMixBuffer.cpp
|
||||
;
|
||||
|
||||
LinkSharedOSLibs mixer.media_addon : be media ;
|
||||
|
||||
@@ -14,7 +14,7 @@ extern "C" _EXPORT BMediaAddOn* make_media_addon(image_id image) {
|
||||
return new AudioMixerAddon(image);
|
||||
}
|
||||
|
||||
// -------------------------------------------------------- //
|
||||
// ------------------------------------------------------- //
|
||||
// ctor/dtor
|
||||
// -------------------------------------------------------- //
|
||||
|
||||
@@ -32,6 +32,7 @@ status_t AudioMixerAddon::InitCheck(
|
||||
}
|
||||
|
||||
int32 AudioMixerAddon::CountFlavors() {
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
||||
@@ -41,27 +42,26 @@ status_t AudioMixerAddon::GetFlavorAt(
|
||||
if(n)
|
||||
return B_ERROR;
|
||||
|
||||
flavor_info* pInfo = new flavor_info;
|
||||
pInfo->internal_id = n;
|
||||
pInfo->name = "AudioMixer";
|
||||
pInfo->info =
|
||||
flavor_info* fInfo = new flavor_info;
|
||||
fInfo->internal_id = n;
|
||||
fInfo->name = "AudioMixer";
|
||||
fInfo->info =
|
||||
"AudioMixer media addon\n";
|
||||
"By David Shipman, 2002";
|
||||
pInfo->kinds = B_BUFFER_PRODUCER | B_BUFFER_CONSUMER | B_SYSTEM_MIXER | B_CONTROLLABLE;
|
||||
pInfo->flavor_flags = 0;
|
||||
pInfo->possible_count = 0;
|
||||
fInfo->kinds = B_BUFFER_PRODUCER | B_BUFFER_CONSUMER | B_SYSTEM_MIXER | B_CONTROLLABLE;
|
||||
fInfo->flavor_flags = B_FLAVOR_IS_LOCAL;
|
||||
fInfo->possible_count = 500;
|
||||
|
||||
media_format* pFormat = new media_format;
|
||||
pFormat->type = B_MEDIA_RAW_AUDIO;
|
||||
pFormat->u.raw_audio = media_raw_audio_format::wildcard;
|
||||
media_format* fFormat = new media_format;
|
||||
fFormat->type = B_MEDIA_RAW_AUDIO;
|
||||
fFormat->u.raw_audio = media_raw_audio_format::wildcard;
|
||||
|
||||
pInfo->in_format_count = 1;
|
||||
pInfo->in_formats = pFormat;
|
||||
fInfo->in_format_count = 1;
|
||||
fInfo->in_formats = fFormat;
|
||||
|
||||
pInfo->out_format_count = 1;
|
||||
pInfo->out_formats = pFormat;
|
||||
fInfo->out_format_count = 1;
|
||||
fInfo->out_formats = fFormat;
|
||||
|
||||
*out_info = pInfo;
|
||||
*out_info = fInfo;
|
||||
return B_OK;
|
||||
}
|
||||
|
||||
@@ -78,7 +78,7 @@ status_t AudioMixerAddon::GetConfigurationFor(
|
||||
BMessage* into_message) {
|
||||
|
||||
// no config yet
|
||||
return B_OK;
|
||||
return B_ERROR;
|
||||
}
|
||||
|
||||
// end
|
||||
@@ -1,39 +1,35 @@
|
||||
OpenBeOS Audio Mixer
|
||||
David Shipman, 2002
|
||||
David Shipman, 14/08/2002
|
||||
|
||||
Overview
|
||||
|
||||
The node is based on a mediaeventlooper, using the HandleEvent loop to compile
|
||||
the mixed buffer output.
|
||||
Each input creates a ringbuffer (fixed size "looped" buffer) for its input - this way
|
||||
buffer contents can be placed in the ringbuffer and recycled immediately.
|
||||
The node is based on a mediaeventlooper, using the HandleEvent loop to compile the mixed buffer output.
|
||||
Each input creates a ringbuffer (fixed size "looped" buffer) for its input - this way buffer contents can be
|
||||
placed in the ringbuffer and the buffer recycled immediately.
|
||||
|
||||
Inputs are maintained using a list of input_channel structs - inputs are created/
|
||||
destroyed dynamically when producers connect/disconnect.
|
||||
Inputs are maintained using a list of mixer_input objects - inputs are created/destroyed dynamically when
|
||||
producers connect/disconnect.
|
||||
|
||||
Done
|
||||
|
||||
- node functions as a replacement for the BeOS R5 mixer.media-addon
|
||||
- IO/conversion between stereo B_AUDIO_FLOAT and B_AUDIO_SHORT streams
|
||||
- IO/conversion between the major audio types (FLOAT, SHORT)
|
||||
- interface mostly done (all gain controls operational)
|
||||
- mixing with different gain levels is functional
|
||||
|
||||
Tested
|
||||
|
||||
- Output to emu10k1 (SBLive)
|
||||
- Input from CL-Amp and emu10k1-in
|
||||
- Input from CL-Amp, file-readers, SoundPlay, music software - almost all work well
|
||||
|
||||
Bugs
|
||||
|
||||
- Soundplay, extractor nodes, and a variety of others connect, but sound really bad!
|
||||
- When input stops, the current ringbuffer contents continue to be played
|
||||
|
||||
To Do (lots) (vaguely in order of importance)
|
||||
To Do (vaguely in order of importance)
|
||||
|
||||
- fix bugs
|
||||
- support for remaining formats
|
||||
- samplerate conversion
|
||||
- Interface (BControllable)
|
||||
- format negotation fine-tuning - some nodes still connect with weird formats
|
||||
- rewrite of buffer mixing routines - at the moment its really inefficient, and a total mess
|
||||
- complete interface (add mutes, panning)
|
||||
- multithreading (separate mix thread)
|
||||
- tidy up code
|
||||
- mixer save (to save parameter states when a node is disconnected)
|
||||
|
||||
Notes :
|
||||
|
||||
|
||||
Reference in New Issue
Block a user