mirror of
https://github.com/WinampDesktop/winamp.git
synced 2024-09-24 15:54:12 +00:00
388 lines
9.7 KiB
C++
388 lines
9.7 KiB
C++
#pragma once
|
|
#include <bfc/platform/types.h>
|
|
#include "../Winamp/in2.h"
|
|
#include "../Winamp/out.h"
|
|
#include "SpillBuffer.h"
|
|
#include <assert.h>
|
|
|
|
/* A class to manage Winamp input plugin audio output
|
|
** It handles the following for you:
|
|
** * Ensuring that Vis data is sent in chunks of 576
|
|
** * Dealing with gapless audio
|
|
** (you need to pass in the number of pre-delay and post-delay samples)
|
|
** * dealing with the DSP plugin
|
|
** * Waiting for CanWrite()
|
|
** * dealing with inter-timestamps
|
|
** e.g. you pass it >576 samples and it can give you a timestamp based on the divided chunk position
|
|
|
|
to use, you need to derive from a class that declares
|
|
int WaitOrAbort(int time_in_ms);
|
|
return 0 on success, non-zero when you need to abort. the return value is passed back through Write()
|
|
*/
|
|
|
|
namespace nu // namespace it since "AudioOutput" isn't a unique enough name
|
|
{
|
|
template <class wait_t>
|
|
class AudioOutput : public wait_t
|
|
{
|
|
public:
|
|
AudioOutput( In_Module *plugin ) : plugin( plugin )
|
|
{
|
|
Init( nullptr );
|
|
}
|
|
|
|
~AudioOutput()
|
|
{
|
|
post_buffer.reset();
|
|
buffer576.reset();
|
|
}
|
|
|
|
/* Initializes and sets the output plugin pointer
|
|
** for most input plugins, the nu::AudioOutput object will be a global,
|
|
** so this will be necessary to call at the start of Play thread */
|
|
void Init( Out_Module *_output )
|
|
{
|
|
output = _output;
|
|
audio_opened = false;
|
|
first_timestamp = 0;
|
|
sample_size = 0;
|
|
output_latency = 0;
|
|
|
|
post_buffer.reset();
|
|
buffer576.reset();
|
|
|
|
cut_size = 0;
|
|
pre_cut_size = 0;
|
|
pre_cut = 0;
|
|
decoder_delay = 0;
|
|
channels = 0;
|
|
sample_rate = 0;
|
|
bps = 0;
|
|
}
|
|
|
|
/* sets end-of-stream delay (in samples)
|
|
** WITHOUT componesating for post-delay.
|
|
** some filetypes (e.g. iTunes MP4) store gapless info this way */
|
|
void SetPostDelay(int postSize)
|
|
{
|
|
if (postSize < 0)
|
|
{
|
|
postSize = 0;
|
|
}
|
|
else if (postSize)
|
|
{
|
|
if (sample_size)
|
|
post_buffer.reserve(postSize*sample_size);
|
|
|
|
cut_size = postSize;
|
|
}
|
|
}
|
|
|
|
/* set end-of-stream zero padding, in samples
|
|
** compensates for decoder delay */
|
|
void SetZeroPadding(int postSize)
|
|
{
|
|
postSize -= decoder_delay;
|
|
if (postSize < 0)
|
|
{
|
|
postSize = 0;
|
|
}
|
|
SetPostDelay(postSize);
|
|
}
|
|
|
|
/* set decoder delay, initial zero samples and end-of-stream zero samples, all in one shot
|
|
** adjusts zero samples for decoder delay. call SetDelays() if your zero samples are already compensated */
|
|
void SetGapless(int decoderDelaySize, int preSize, int postSize)
|
|
{
|
|
decoder_delay = decoderDelaySize;
|
|
SetZeroPadding(postSize);
|
|
|
|
pre_cut_size = preSize;
|
|
pre_cut = pre_cut_size + decoder_delay;
|
|
}
|
|
|
|
/* set decoder delay, initial delay and end-of-stream delay, all in one shot
|
|
** WITHOUT componesating for post-delay.
|
|
** some filetypes (e.g. iTunes MP4) store gapless info this way */
|
|
void SetDelays(int decoderDelaySize, int preSize, int postSize)
|
|
{
|
|
decoder_delay = decoderDelaySize;
|
|
SetPostDelay(postSize);
|
|
|
|
pre_cut_size = preSize;
|
|
pre_cut = pre_cut_size;
|
|
}
|
|
|
|
/* Call on seek */
|
|
void Flush(int time_in_ms)
|
|
{
|
|
if (audio_opened)
|
|
{
|
|
pre_cut = pre_cut_size;
|
|
|
|
output->Flush(time_in_ms);
|
|
first_timestamp = 0; // once we've flushed, we should be accurate so no need for this anymore
|
|
buffer576.clear();
|
|
post_buffer.clear();
|
|
}
|
|
else
|
|
first_timestamp = time_in_ms;
|
|
}
|
|
|
|
bool Opened() const
|
|
{
|
|
return audio_opened;
|
|
}
|
|
|
|
int GetLatency() const
|
|
{
|
|
return output_latency;
|
|
}
|
|
|
|
int GetFirstTimestamp() const
|
|
{
|
|
return first_timestamp;
|
|
}
|
|
|
|
/* timestamp is meant to be the first timestamp according to the containing file format
|
|
** e.g. many MP4 videos start on 12ms or something, for accurate a/v syncing */
|
|
bool Open(int timestamp, int channels, int sample_rate, int bps, int buffer_len_ms=-1, int pre_buffer_ms=-1)
|
|
{
|
|
if (!audio_opened)
|
|
{
|
|
int latency = output->Open(sample_rate, channels, bps, buffer_len_ms, pre_buffer_ms);
|
|
if (latency < 0)
|
|
return false;
|
|
plugin->SAVSAInit(latency, sample_rate);
|
|
plugin->VSASetInfo(sample_rate, channels);
|
|
output->SetVolume(-666);
|
|
plugin->SetInfo(-1, sample_rate / 1000, channels, /* TODO? 0*/1);
|
|
|
|
output_latency = latency;
|
|
first_timestamp = timestamp;
|
|
sample_size = channels*bps / 8;
|
|
this->channels=channels;
|
|
this->sample_rate=sample_rate;
|
|
this->bps=bps;
|
|
SetPostDelay((int)cut_size); // set this again now that we know sample_size, so buffers get allocated correctly
|
|
buffer576.reserve(576*sample_size);
|
|
audio_opened=true;
|
|
}
|
|
return audio_opened;
|
|
}
|
|
|
|
void Close()
|
|
{
|
|
if (audio_opened && output)
|
|
{
|
|
output->Close();
|
|
plugin->SAVSADeInit();
|
|
}
|
|
output = 0;
|
|
first_timestamp = 0;
|
|
}
|
|
|
|
/* outSize is in bytes
|
|
** */
|
|
int Write(char *out, size_t outSize)
|
|
{
|
|
if (!out && !outSize)
|
|
{
|
|
/* --- write contents of buffered audio (end-zero-padding buffer) */
|
|
if (!post_buffer.empty())
|
|
{
|
|
void *buffer = 0;
|
|
size_t len = 0;
|
|
if (post_buffer.get(&buffer, &len))
|
|
{
|
|
int ret = Write576((char *)buffer, len);
|
|
if (ret != 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* --- write any remaining data in 576 spill buffer (skip vis) */
|
|
if (!buffer576.empty())
|
|
{
|
|
void *buffer = 0;
|
|
size_t len = 0;
|
|
if (buffer576.get(&buffer, &len))
|
|
{
|
|
int ret = WriteOutput((char *)buffer, len);
|
|
if (ret != 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
output->Write(0, 0);
|
|
return 0;
|
|
}
|
|
|
|
// this probably should not happen but have seen it in some crash reports
|
|
if (!sample_size)
|
|
return 0;
|
|
|
|
assert((outSize % sample_size) == 0);
|
|
size_t outSamples = outSize / sample_size;
|
|
|
|
/* --- cut pre samples, if necessary --- */
|
|
size_t pre = min(pre_cut, outSamples);
|
|
out += pre * sample_size;
|
|
outSize -= pre * sample_size;
|
|
pre_cut -= pre;
|
|
//outSize = outSamples * sample_size;
|
|
|
|
// do we will have samples to output after cutting pre-delay?
|
|
if (!outSize)
|
|
return 0;
|
|
|
|
/* --- if we don't have enough to fully fill the end-zero-padding buffer, go ahead and fill --- */
|
|
if (outSize < post_buffer.length())
|
|
{
|
|
size_t bytes_written = post_buffer.write(out, outSize);
|
|
out+=bytes_written;
|
|
outSize-=bytes_written;
|
|
}
|
|
|
|
// if we're out of samples, go ahead and bail
|
|
if (!outSize)
|
|
return 0;
|
|
|
|
/* --- write contents of buffered audio (end-zero-padding buffer) */
|
|
if (!post_buffer.empty())
|
|
{
|
|
void *buffer = 0;
|
|
size_t len = 0;
|
|
if (post_buffer.get(&buffer, &len))
|
|
{
|
|
int ret = Write576((char *)buffer, len);
|
|
if (ret != 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* --- make sure we have enough samples left over to fill our post-zero-padding buffer --- */
|
|
size_t remainingFill = /*cut_size - */post_buffer.remaining();
|
|
int outWrite = max(0, (int)outSize - (int)remainingFill);
|
|
|
|
/* --- write the output that doesn't end up in the post buffer */
|
|
if (outWrite)
|
|
{
|
|
int ret = Write576(out, outWrite);
|
|
if (ret != 0)
|
|
return ret;
|
|
}
|
|
out += outWrite;
|
|
outSize -= outWrite;
|
|
|
|
/* --- write whatever is left over into the end-zero-padding buffer --- */
|
|
if (outSize)
|
|
{
|
|
post_buffer.write(out, outSize);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* meant to be called after Write(0,0) */
|
|
int WaitWhilePlaying()
|
|
{
|
|
while (output->IsPlaying())
|
|
{
|
|
int ret = WaitOrAbort(10);
|
|
if (ret != 0)
|
|
return ret;
|
|
|
|
output->CanWrite(); // some output drivers need CanWrite
|
|
// to be called on a regular basis.
|
|
}
|
|
return 0;
|
|
}
|
|
private:
|
|
/* helper methods */
|
|
int WaitForOutput(int write_size_bytes)
|
|
{
|
|
while (output->CanWrite() < write_size_bytes)
|
|
{
|
|
int ret = WaitOrAbort(55);
|
|
if (ret != 0)
|
|
return ret;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
/* writes one chunk (576 samples) to the output plugin, waiting as necessary */
|
|
int WriteOutput(char *buffer, size_t len)
|
|
{
|
|
int ret = WaitForOutput((int)len);
|
|
if (ret != 0)
|
|
return ret;
|
|
|
|
// write vis data before so we guarantee 576 samples
|
|
if (len == 576*sample_size)
|
|
{
|
|
plugin->SAAddPCMData(buffer, channels, bps, output->GetWrittenTime() + first_timestamp);
|
|
plugin->VSAAddPCMData(buffer, channels, bps, output->GetWrittenTime() + first_timestamp);
|
|
}
|
|
|
|
if (plugin->dsp_isactive())
|
|
len = sample_size * plugin->dsp_dosamples((short *)buffer, (int)(len / sample_size), bps, channels, sample_rate);
|
|
|
|
output->Write(buffer, (int)len);
|
|
return 0;
|
|
}
|
|
|
|
/* given a large buffer, writes 576 sample chunks to the vis, dsp and output plugin */
|
|
int Write576(char *buffer, size_t out_size)
|
|
{
|
|
/* if we have some stuff leftover in the 576 sample spill buffer, fill it up */
|
|
if (!buffer576.empty())
|
|
{
|
|
size_t bytes_written = buffer576.write(buffer, out_size);
|
|
out_size -= bytes_written;
|
|
buffer += bytes_written;
|
|
}
|
|
|
|
if (buffer576.full())
|
|
{
|
|
void *buffer = 0;
|
|
size_t len = 0;
|
|
if (buffer576.get(&buffer, &len))
|
|
{
|
|
int ret = WriteOutput((char *)buffer, len);
|
|
if (ret != 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
while (out_size >= 576*sample_size)
|
|
{
|
|
int ret = WriteOutput(buffer, 576*sample_size);
|
|
if (ret != 0)
|
|
return ret;
|
|
|
|
out_size -= 576*sample_size;
|
|
buffer+=576*sample_size;
|
|
}
|
|
|
|
if (out_size)
|
|
{
|
|
assert(out_size < 576*sample_size);
|
|
buffer576.write(buffer, out_size);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
private:
|
|
Out_Module *output;
|
|
In_Module *plugin;
|
|
SpillBuffer post_buffer, buffer576;
|
|
size_t cut_size;
|
|
size_t pre_cut, pre_cut_size, decoder_delay;
|
|
bool audio_opened;
|
|
int first_timestamp; /* timestamp of the first decoded audio frame, necessary for accurate video syncing */
|
|
size_t sample_size; /* size, in bytes, of one sample of audio (channels*bps/8) */
|
|
int output_latency; /* as returned from Out_Module::Open() */
|
|
int channels, sample_rate, bps;
|
|
};
|
|
}
|