Aegisub/core/audio_provider.cpp
Rodrigo Braz Monteiro bd116bd286 Changing vertical zoom will now also change the playback volume
Originally committed to SVN as r88.
2006-02-20 23:54:15 +00:00

606 lines
14 KiB
C++

// Copyright (c) 2005, Rodrigo Braz Monteiro
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// -----------------------------------------------------------------------------
//
// AEGISUB
//
// Website: http://aegisub.cellosoft.com
// Contact: mailto:zeratul@cellosoft.com
//
///////////
// Headers
#include <wx/wxprec.h>
#include <wx/filename.h>
#include <Mmreg.h>
#include "avisynth_wrap.h"
#include "utils.h"
#include "audio_provider.h"
#include "video_display.h"
#include "options.h"
#include "audio_display.h"
#include "main.h"
#include "dialog_progress.h"
extern "C" {
#include <portaudio.h>
}
int AudioProvider::pa_refcount = 0;
#define CacheBits ((22))
#define CacheBlockSize ((1 << CacheBits))
//////////////
// Constructor
AudioProvider::AudioProvider(wxString _filename, AudioDisplay *_display) {
type = AUDIO_PROVIDER_NONE;
playing = false;
stopping = false;
blockcache = NULL;
blockcount = 0;
raw = NULL;
display = _display;
blockcount = 0;
volume = 1.0f;
filename = _filename;
// Initialize portaudio
if (!pa_refcount) {
PaError err = Pa_Initialize();
if (err != paNoError)
throw _T("Failed opening PortAudio with error: ") + wxString(Pa_GetErrorText(err),wxConvLocal);
pa_refcount++;
}
try {
OpenAVSAudio();
OpenStream();
} catch (...) {
Unload();
throw;
}
}
//////////////
// Destructor
AudioProvider::~AudioProvider() {
CloseStream();
Unload();
}
////////////////
// Unload audio
void AudioProvider::Unload() {
// Clean up avisynth
clip = NULL;
// Close file
if (type == AUDIO_PROVIDER_DISK_CACHE) {
file_cache.close();
wxRemoveFile(DiskCacheName());
}
// Free ram cache
if (blockcache) {
for (int i = 0; i < blockcount; i++)
if (blockcache[i])
delete blockcache[i];
delete blockcache;
}
// Clear buffers
delete raw;
if (!--pa_refcount)
Pa_Terminate();
}
////////////////////////////
// Load audio from avisynth
void AudioProvider::OpenAVSAudio() {
// Set variables
type = AUDIO_PROVIDER_AVS;
AVSValue script;
// Prepare avisynth
wxMutexLocker lock(AviSynthMutex);
try {
const char * argnames[3] = { 0, "video", "audio" };
AVSValue args[3] = { env->SaveString(filename.mb_str(wxConvLocal)), false, true };
script = env->Invoke("DirectShowSource", AVSValue(args,3),argnames);
LoadFromClip(script);
} catch (AvisynthError &err) {
throw _T("AviSynth error: ") + wxString(err.msg,wxConvLocal);
}
}
/////////////////////////
// Read from environment
void AudioProvider::LoadFromClip(AVSValue _clip) {
// Prepare avisynth
AVSValue script;
// Check if it has audio
VideoInfo vi = _clip.AsClip()->GetVideoInfo();
if (!vi.HasAudio()) throw wxString(_T("No audio found."));
// Convert to one channel
char buffer[1024];
strcpy(buffer,Options.AsText(_T("Audio Downmixer")).mb_str(wxConvLocal));
script = env->Invoke(buffer, _clip);
// Convert to 16 bits per sample
script = env->Invoke("ConvertAudioTo16bit", script);
// Convert sample rate
int setsample = Options.AsInt(_T("Audio Sample Rate"));
if (setsample != 0) {
AVSValue args[2] = { script, setsample };
script = env->Invoke("ResampleAudio", AVSValue(args,2));
}
// Set clip
PClip tempclip = script.AsClip();
vi = tempclip->GetVideoInfo();
// Read properties
channels = vi.AudioChannels();
num_samples = vi.num_audio_samples;
sample_rate = vi.SamplesPerSecond();
bytes_per_sample = vi.BytesPerAudioSample();
// Read whole thing into ram cache
if (Options.AsInt(_T("Audio Cache")) == 1) {
ConvertToRAMCache(tempclip);
clip = NULL;
}
// Disk cache
else if (Options.AsInt(_T("Audio Cache")) == 2) {
ConvertToDiskCache(tempclip);
clip = NULL;
}
// Assign to avisynth
else {
clip = tempclip;
}
}
/////////////
// RAM Cache
void AudioProvider::ConvertToRAMCache(PClip &tempclip) {
// Allocate cache
__int64 ssize = num_samples * bytes_per_sample;
blockcount = (ssize + CacheBlockSize - 1) >> CacheBits;
blockcache = new char*[blockcount];
for (int i = 0; i < blockcount; i++)
blockcache[i] = NULL;
try {
for (int i = 0; i < blockcount; i++)
blockcache[i] = new char[MIN(CacheBlockSize,ssize-i*CacheBlockSize)];
} catch (...) {
for (int i = 0; i < blockcount; i++)
delete blockcache[i];
delete blockcache;
blockcache = NULL;
blockcount = 0;
if (wxMessageBox(_("Not enough ram available. Use disk cache instead?"),_("Audio Information"),wxICON_INFORMATION | wxYES_NO) == wxYES) {
ConvertToDiskCache(tempclip);
return;
} else
throw wxString(_T("Couldn't open audio, not enough ram available."));
}
// Start progress
volatile bool canceled = false;
DialogProgress *progress = new DialogProgress(NULL,_("Load audio"),&canceled,_("Reading into RAM"),0,num_samples);
progress->Show();
progress->SetProgress(0,1);
// Read cache
int readsize = CacheBlockSize / bytes_per_sample;
for (int i=0;i<blockcount && !canceled; i++) {
tempclip->GetAudio((char*)blockcache[i],i*readsize, i == blockcount-1 ? (num_samples - i*readsize) : readsize,env);
progress->SetProgress(i,blockcount-1);
}
type = AUDIO_PROVIDER_CACHE;
// Clean up progress
if (!canceled)
progress->Destroy();
else
throw wxString(_T("Audio loading cancelled by user"));
}
//////////////
// Disk Cache
void AudioProvider::ConvertToDiskCache(PClip &tempclip) {
// Check free space
wxLongLong freespace;
if (wxGetDiskSpace(DiskCachePath(), NULL, &freespace))
if (num_samples * channels * bytes_per_sample > freespace)
throw wxString(_T("Not enough free diskspace in "))+DiskCachePath()+wxString(_T(" to cache the audio"));
// Open output file
std::ofstream file;
char filename[512];
strcpy(filename,DiskCacheName().mb_str(wxConvLocal));
file.open(filename,std::ios::binary | std::ios::out | std::ios::trunc);
// Start progress
volatile bool canceled = false;
DialogProgress *progress = new DialogProgress(NULL,_T("Load audio"),&canceled,_T("Reading to Hard Disk cache"),0,num_samples);
progress->Show();
// Write to disk
int block = 4096;
char *temp = new char[block * channels * bytes_per_sample];
for (__int64 i=0;i<num_samples && !canceled; i+=block) {
if (block+i > num_samples) block = num_samples - i;
tempclip->GetAudio(temp,i,block,env);
file.write(temp,block * channels * bytes_per_sample);
progress->SetProgress(i,num_samples);
}
file.close();
type = AUDIO_PROVIDER_DISK_CACHE;
// Finish
if (!canceled) {
progress->Destroy();
file_cache.open(filename,std::ios::binary | std::ios::in);
}
else
throw wxString(_T("Audio loading cancelled by user"));
}
////////////////
// Get filename
wxString AudioProvider::GetFilename() {
return filename;
}
int aaa = 0;
/////////////
// Get audio
void AudioProvider::GetAudio(void *buf, __int64 start, __int64 count) {
// Requested beyond the length of audio
if (start+count > num_samples) {
__int64 oldcount = count;
count = num_samples-start;
if (count < 0) count = 0;
// Fill beyond with zero
if (bytes_per_sample == 1) {
char *temp = (char *) buf;
for (int i=count;i<oldcount;i++) {
temp[i] = 0;
}
}
if (bytes_per_sample == 2) {
short *temp = (short *) buf;
for (int i=count;i<oldcount;i++) {
temp[i] = 0;
}
}
}
if (count) {
char *charbuf = (char *)buf;
// RAM Cache
if (type == AUDIO_PROVIDER_CACHE) {
int i = (start*bytes_per_sample) >> CacheBits;
int start_offset = (start*bytes_per_sample) & (CacheBlockSize-1);
__int64 bytesremaining = count*bytes_per_sample;
while (bytesremaining) {
int readsize=MIN(bytesremaining,CacheBlockSize);
readsize = MIN(readsize,CacheBlockSize - start_offset);
memcpy(charbuf,(char *)(blockcache[i++]+start_offset),readsize);
charbuf+=readsize;
start_offset=0;
bytesremaining-=readsize;
}
}
// Disk cache
else if (type == AUDIO_PROVIDER_DISK_CACHE) {
wxMutexLocker disklock(diskmutex);
file_cache.seekg(start*bytes_per_sample);
file_cache.read((char*)buf,count*bytes_per_sample*channels);
}
// Avisynth
else {
wxMutexLocker disklock(diskmutex);
clip->GetAudio(buf,start,count,env);
}
}
}
//////////////////////////
// Get number of channels
int AudioProvider::GetChannels() {
return channels;
}
//////////////////////////
// Get number of samples
__int64 AudioProvider::GetNumSamples() {
return num_samples;
}
///////////////////
// Get sample rate
int AudioProvider::GetSampleRate() {
return sample_rate;
}
////////////////
// Get waveform
void AudioProvider::GetWaveForm(int *min,int *peak,__int64 start,int w,int h,int samples,float scale) {
// Setup
int channels = GetChannels();
int n = w * samples;
for (int i=0;i<w;i++) {
peak[i] = 0;
min[i] = h;
}
// Prepare waveform
int cur;
int curvalue;
// Prepare buffers
int needLen = n*channels*bytes_per_sample;
if (raw) {
if (raw_len < needLen) {
delete raw;
raw = NULL;
}
}
if (!raw) {
raw_len = needLen;
raw = (void*) new char[raw_len];
}
if (bytes_per_sample == 1) {
// Read raw samples
unsigned char *raw_char = (unsigned char*) raw;
GetAudio(raw,start,n);
int amplitude = h*scale;
// Calculate waveform
for (int i=0;i<n;i++) {
cur = i/samples;
curvalue = h - (int(raw_char[i*channels])*amplitude)/0xFF;
if (curvalue > h) curvalue = h;
if (curvalue < 0) curvalue = 0;
if (curvalue < min[cur]) min[cur] = curvalue;
if (curvalue > peak[cur]) peak[cur] = curvalue;
}
}
if (bytes_per_sample == 2) {
// Read raw samples
short *raw_short = (short*) raw;
GetAudio(raw,start,n);
int half_h = h/2;
int half_amplitude = half_h * scale;
// Calculate waveform
for (int i=0;i<n;i++) {
cur = i/samples;
curvalue = half_h - (int(raw_short[i*channels])*half_amplitude)/0x8000;
if (curvalue > h) curvalue = h;
if (curvalue < 0) curvalue = 0;
if (curvalue < min[cur]) min[cur] = curvalue;
if (curvalue > peak[cur]) peak[cur] = curvalue;
}
}
}
///////////////////////////
// Get disk cache path
wxString AudioProvider::DiskCachePath() {
return AegisubApp::folderName;
}
///////////////////////////
// Get disk cache filename
wxString AudioProvider::DiskCacheName() {
return DiskCachePath() + _T("audio.tmp");
}
//////////////////////
// PortAudio callback
int paCallback(void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, PaTimestamp outTime, void *userData) {
// Get provider
AudioProvider *provider = (AudioProvider *) userData;
int end = 0;
// Calculate how much left
__int64 lenAvailable = provider->endPos - provider->playPos;
unsigned __int64 avail = 0;
if (lenAvailable > 0) {
avail = lenAvailable;
if (avail > framesPerBuffer) {
lenAvailable = framesPerBuffer;
avail = lenAvailable;
}
}
else {
lenAvailable = 0;
avail = 0;
}
// Play something
if (lenAvailable > 0) {
provider->GetAudio(outputBuffer,provider->playPos,lenAvailable);
}
// Pad end with blank
if (avail < (unsigned __int64) framesPerBuffer) {
provider->softStop = true;
}
// Set volume
short *output = (short*) outputBuffer;
for (unsigned int i=0;i<avail;i++) output[i] = MID(-(1<<15),int(output[i] * provider->volume),(1<<15)-1);
// Fill rest with blank
for (unsigned int i=avail;i<framesPerBuffer;i++) output[i]=0;
// Set play position (and real one)
provider->playPos += framesPerBuffer;
provider->realPlayPos = provider->playPos - (outTime - Pa_StreamTime(provider->stream));
// Cap to start if lower
return end;
}
////////
// Play
void AudioProvider::Play(__int64 start,__int64 count) {
// Stop if it's already playing
wxMutexLocker locker(PAMutex);
// Set values
endPos = start + count;
realPlayPos = start;
playPos = start;
startPos = start;
startMS = startPos * 1000 / GetSampleRate();
// Start playing
if (!playing) {
PaError err = Pa_StartStream(stream);
if (err != paNoError) {
return;
}
}
playing = true;
if (!display->UpdateTimer.IsRunning()) display->UpdateTimer.Start(15);
}
////////
// Stop
void AudioProvider::Stop(bool timerToo) {
//wxMutexLocker locker(PAMutex);
softStop = false;
// Stop stream
playing = false;
Pa_StopStream (stream);
realPlayPos = 0;
// Stop timer
if (timerToo) {
display->UpdateTimer.Stop();
}
}
///////////////
// Open stream
void AudioProvider::OpenStream() {
// Open stream
PaError err = Pa_OpenDefaultStream(&stream,0,GetChannels(),paInt16,GetSampleRate(),256,16,paCallback,this);
if (err != paNoError)
throw wxString(_T("Failed initializing PortAudio stream with error: ") + wxString(Pa_GetErrorText(err),wxConvLocal));
}
///////////////
// Close stream
void AudioProvider::CloseStream() {
try {
Stop(false);
Pa_CloseStream(stream);
} catch (...) {}
}
/////////////////////
// Ask to stop later
void AudioProvider::RequestStop() {
wxCommandEvent event(wxEVT_STOP_AUDIO, 1000);
event.SetEventObject(this);
wxMutexGuiEnter();
AddPendingEvent(event);
wxMutexGuiLeave();
}
/////////
// Event
DEFINE_EVENT_TYPE(wxEVT_STOP_AUDIO)
BEGIN_EVENT_TABLE(AudioProvider, wxEvtHandler)
EVT_COMMAND (1000, wxEVT_STOP_AUDIO, AudioProvider::OnStopAudio)
END_EVENT_TABLE()
void AudioProvider::OnStopAudio(wxCommandEvent &event) {
Stop(false);
}