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-rw-r--r--examples/alstreamcb.cpp145
1 files changed, 51 insertions, 94 deletions
diff --git a/examples/alstreamcb.cpp b/examples/alstreamcb.cpp
index fbc3d02d..8fb1f102 100644
--- a/examples/alstreamcb.cpp
+++ b/examples/alstreamcb.cpp
@@ -36,9 +36,7 @@
#include <thread>
#include <vector>
-#include "SDL_sound.h"
-#include "SDL_audio.h"
-#include "SDL_stdinc.h"
+#include "sndfile.h"
#include "AL/al.h"
#include "AL/alc.h"
@@ -46,14 +44,6 @@
#include "common/alhelpers.h"
-#ifndef SDL_AUDIO_MASK_BITSIZE
-#define SDL_AUDIO_MASK_BITSIZE (0xFF)
-#endif
-#ifndef SDL_AUDIO_BITSIZE
-#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
-#endif
-
-
#ifndef AL_SOFT_callback_buffer
#define AL_SOFT_callback_buffer
typedef unsigned int ALbitfieldSOFT;
@@ -87,13 +77,12 @@ struct StreamPlayer {
size_t mStartOffset{0};
/* Handle for the audio file to decode. */
- Sound_Sample *mSample{nullptr};
- Uint32 mAvailableData{0};
+ SNDFILE *mSndfile{nullptr};
+ SF_INFO mSfInfo{};
size_t mDecoderOffset{0};
/* The format of the callback samples. */
ALenum mFormat;
- ALsizei mSampleRate;
StreamPlayer()
{
@@ -111,18 +100,18 @@ struct StreamPlayer {
{
alDeleteSources(1, &mSource);
alDeleteBuffers(1, &mBuffer);
- if(mSample)
- Sound_FreeSample(mSample);
+ if(mSndfile)
+ sf_close(mSndfile);
}
void close()
{
- if(mSample)
+ if(mSndfile)
{
alSourceRewind(mSource);
alSourcei(mSource, AL_BUFFER, 0);
- Sound_FreeSample(mSample);
- mSample = nullptr;
+ sf_close(mSndfile);
+ mSndfile = nullptr;
}
}
@@ -130,50 +119,30 @@ struct StreamPlayer {
{
close();
- /* Open the file in its normal format. */
- mSample = Sound_NewSampleFromFile(filename, nullptr, 0);
- if(!mSample)
+ /* Open the file and figure out the OpenAL format. */
+ mSndfile = sf_open(filename, SFM_READ, &mSfInfo);
+ if(!mSndfile)
{
- fprintf(stderr, "Could not open audio in %s\n", filename);
+ fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(mSndfile));
return false;
}
- /* Figure out the OpenAL format from the sample's format. */
mFormat = AL_NONE;
- if(mSample->actual.channels == 1)
- {
- if(mSample->actual.format == AUDIO_U8)
- mFormat = AL_FORMAT_MONO8;
- else if(mSample->actual.format == AUDIO_S16SYS)
- mFormat = AL_FORMAT_MONO16;
- }
- else if(mSample->actual.channels == 2)
- {
- if(mSample->actual.format == AUDIO_U8)
- mFormat = AL_FORMAT_STEREO8;
- else if(mSample->actual.format == AUDIO_S16SYS)
- mFormat = AL_FORMAT_STEREO16;
- }
- if(!mFormat)
+ if(mSfInfo.channels == 1)
+ mFormat = AL_FORMAT_MONO16;
+ else if(mSfInfo.channels == 2)
+ mFormat = AL_FORMAT_STEREO16;
+ else
{
- fprintf(stderr, "Unsupported sample format: 0x%04x, %d channels\n",
- mSample->actual.format, mSample->actual.channels);
- Sound_FreeSample(mSample);
- mSample = nullptr;
+ fprintf(stderr, "Unsupported channel count: %d\n", mSfInfo.channels);
+ sf_close(mSndfile);
+ mSndfile = nullptr;
return false;
}
- mSampleRate = static_cast<ALsizei>(mSample->actual.rate);
-
- const auto frame_size = Uint32{mSample->actual.channels} *
- SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
-
- /* Set a 50ms decode buffer size. */
- Sound_SetBufferSize(mSample, static_cast<Uint32>(mSampleRate)*50/1000 * frame_size);
- mAvailableData = 0;
/* Set a 1s ring buffer size. */
- mBufferDataSize = static_cast<Uint32>(mSampleRate) * size_t{frame_size};
+ mBufferDataSize = static_cast<ALuint>(mSfInfo.samplerate*mSfInfo.channels) * sizeof(short);
mBufferData.reset(new ALbyte[mBufferDataSize]);
mReadPos.store(0, std::memory_order_relaxed);
mWritePos.store(0, std::memory_order_relaxed);
@@ -239,34 +208,27 @@ struct StreamPlayer {
bool prepare()
{
- alBufferCallbackSOFT(mBuffer, mFormat, mSampleRate, bufferCallbackC, this, 0);
+ alBufferCallbackSOFT(mBuffer, mFormat, mSfInfo.samplerate, bufferCallbackC, this, 0);
alSourcei(mSource, AL_BUFFER, static_cast<ALint>(mBuffer));
if(ALenum err{alGetError()})
{
fprintf(stderr, "Failed to set callback: %s (0x%04x)\n", alGetString(err), err);
return false;
}
-
- mAvailableData = Sound_Decode(mSample);
- if(!mAvailableData)
- fprintf(stderr, "Failed to decode any samples: %s\n", Sound_GetError());
- return mAvailableData != 0;
+ return true;
}
bool update()
{
- constexpr int BadFlags{SOUND_SAMPLEFLAG_EOF | SOUND_SAMPLEFLAG_ERROR};
-
ALenum state;
ALint pos;
alGetSourcei(mSource, AL_SAMPLE_OFFSET, &pos);
alGetSourcei(mSource, AL_SOURCE_STATE, &state);
+ const size_t frame_size{static_cast<ALuint>(mSfInfo.channels) * sizeof(short)};
size_t woffset{mWritePos.load(std::memory_order_acquire)};
if(state != AL_INITIAL)
{
- const auto frame_size = Uint32{mSample->actual.channels} *
- SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
roffset};
@@ -276,15 +238,17 @@ struct StreamPlayer {
* the playback offset the source was started with.
*/
const size_t curtime{((state==AL_STOPPED) ? (mDecoderOffset-readable) / frame_size
- : (static_cast<ALuint>(pos) + mStartOffset/frame_size)) / mSample->actual.rate};
+ : (static_cast<ALuint>(pos) + mStartOffset/frame_size))
+ / static_cast<ALuint>(mSfInfo.samplerate)};
printf("\r%3zus (%3zu%% full)", curtime, readable * 100 / mBufferDataSize);
}
else
fputs("Starting...", stdout);
fflush(stdout);
- while(mAvailableData > 0)
+ while(!sf_error(mSndfile))
{
+ size_t read_bytes;
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
if(roffset > woffset)
{
@@ -294,45 +258,39 @@ struct StreamPlayer {
* instead of full.
*/
const size_t writable{roffset-woffset-1};
- /* Don't copy the sample data if it can't all fit. */
- if(writable < mAvailableData) break;
+ if(writable < frame_size) break;
+
+ sf_count_t num_frames{sf_readf_short(mSndfile,
+ reinterpret_cast<short*>(&mBufferData[woffset]),
+ static_cast<sf_count_t>(writable/frame_size))};
+ if(num_frames < 1) break;
- memcpy(&mBufferData[woffset], mSample->buffer, mAvailableData);
- woffset += mAvailableData;
+ read_bytes = static_cast<size_t>(num_frames) * frame_size;
+ woffset += read_bytes;
}
else
{
/* If the read offset is at or behind the write offset, the
* writeable area (might) wrap around. Make sure the sample
- * data can fit, and calculate how much goes in front and in
- * back.
+ * data can fit, and calculate how much can go in front before
+ * wrapping.
*/
- const size_t writable{mBufferDataSize+roffset-woffset-1};
- if(writable < mAvailableData) break;
+ const size_t writable{!roffset ? mBufferDataSize-woffset-1 :
+ (mBufferDataSize-woffset)};
+ if(writable < frame_size) break;
- const size_t todo1{std::min<size_t>(mAvailableData, mBufferDataSize-woffset)};
- const size_t todo2{mAvailableData - todo1};
+ sf_count_t num_frames{sf_readf_short(mSndfile,
+ reinterpret_cast<short*>(&mBufferData[woffset]),
+ static_cast<sf_count_t>(writable/frame_size))};
+ if(num_frames < 1) break;
- memcpy(&mBufferData[woffset], mSample->buffer, todo1);
- woffset += todo1;
+ read_bytes = static_cast<size_t>(num_frames) * frame_size;
+ woffset += read_bytes;
if(woffset == mBufferDataSize)
- {
woffset = 0;
- if(todo2 > 0)
- {
- memcpy(&mBufferData[woffset], static_cast<ALbyte*>(mSample->buffer)+todo1,
- todo2);
- woffset += todo2;
- }
- }
}
mWritePos.store(woffset, std::memory_order_release);
- mDecoderOffset += mAvailableData;
-
- if(!(mSample->flags&BadFlags))
- mAvailableData = Sound_Decode(mSample);
- else
- mAvailableData = 0;
+ mDecoderOffset += read_bytes;
}
if(state != AL_PLAYING && state != AL_PAUSED)
@@ -364,15 +322,14 @@ struct StreamPlayer {
int main(int argc, char **argv)
{
- /* A simple RAII container for OpenAL and SDL_sound startup and shutdown. */
+ /* A simple RAII container for OpenAL startup and shutdown. */
struct AudioManager {
AudioManager(char ***argv_, int *argc_)
{
if(InitAL(argv_, argc_) != 0)
throw std::runtime_error{"Failed to initialize OpenAL"};
- Sound_Init();
}
- ~AudioManager() { Sound_Quit(); CloseAL(); }
+ ~AudioManager() { CloseAL(); }
};
/* Print out usage if no arguments were specified */
@@ -413,7 +370,7 @@ int main(int argc, char **argv)
namepart = argv[i];
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->mFormat),
- player->mSampleRate);
+ player->mSfInfo.samplerate);
fflush(stdout);
if(!player->prepare())
generated by cgit v1.2.3 (git 2.39.1) at 2025-03-20 16:34:51 +0000