quake2quest/Projects/Android/jni/SupportLibs/openal/examples/alstream.c

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/*
* OpenAL Audio Stream Example
*
* Copyright (c) 2011 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains a relatively simple streaming audio player. */
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <signal.h>
#include <assert.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "alhelpers.h"
#include "alffmpeg.h"
LPALBUFFERSAMPLESSOFT palBufferSamplesSOFT = wrap_BufferSamples;
LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT = NULL;
/* Define the number of buffers and buffer size (in samples) to use. 4 buffers
* with 8192 samples each gives a nice per-chunk size, and lets the queue last
* for almost 3/4ths of a second for a 44.1khz stream. */
#define NUM_BUFFERS 4
#define BUFFER_SIZE 8192
typedef struct StreamPlayer {
/* These are the buffers and source to play out through OpenAL with */
ALuint buffers[NUM_BUFFERS];
ALuint source;
/* Handles for the audio stream */
FilePtr file;
StreamPtr stream;
/* A temporary data buffer for readAVAudioData to write to and pass to
* OpenAL with */
ALbyte *data;
ALsizei datasize;
/* The format of the output stream */
ALenum format;
ALenum channels;
ALenum type;
ALuint rate;
} StreamPlayer;
static StreamPlayer *NewPlayer(void);
static void DeletePlayer(StreamPlayer *player);
static int OpenPlayerFile(StreamPlayer *player, const char *filename);
static void ClosePlayerFile(StreamPlayer *player);
static int StartPlayer(StreamPlayer *player);
static int UpdatePlayer(StreamPlayer *player);
/* Creates a new player object, and allocates the needed OpenAL source and
* buffer objects. Error checking is simplified for the purposes of this
* example, and will cause an abort if needed. */
static StreamPlayer *NewPlayer(void)
{
StreamPlayer *player;
player = malloc(sizeof(*player));
assert(player != NULL);
memset(player, 0, sizeof(*player));
/* Generate the buffers and source */
alGenBuffers(NUM_BUFFERS, player->buffers);
assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
alGenSources(1, &player->source);
assert(alGetError() == AL_NO_ERROR && "Could not create source");
/* Set parameters so mono sources play out the front-center speaker and
* won't distance attenuate. */
alSource3i(player->source, AL_POSITION, 0, 0, -1);
alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
return player;
}
/* Destroys a player object, deleting the source and buffers. No error handling
* since these calls shouldn't fail with a properly-made player object. */
static void DeletePlayer(StreamPlayer *player)
{
ClosePlayerFile(player);
alDeleteSources(1, &player->source);
alDeleteBuffers(NUM_BUFFERS, player->buffers);
if(alGetError() != AL_NO_ERROR)
fprintf(stderr, "Failed to delete object IDs\n");
memset(player, 0, sizeof(*player));
free(player);
}
/* Opens the first audio stream of the named file. If a file is already open,
* it will be closed first. */
static int OpenPlayerFile(StreamPlayer *player, const char *filename)
{
ClosePlayerFile(player);
/* Open the file and get the first stream from it */
player->file = openAVFile(filename);
player->stream = getAVAudioStream(player->file, 0);
if(!player->stream)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
goto error;
}
/* Get the stream format, and figure out the OpenAL format */
if(getAVAudioInfo(player->stream, &player->rate, &player->channels,
&player->type) != 0)
{
fprintf(stderr, "Error getting audio info for %s\n", filename);
goto error;
}
player->format = GetFormat(player->channels, player->type, palIsBufferFormatSupportedSOFT);
if(player->format == 0)
{
fprintf(stderr, "Unsupported format (%s, %s) for %s\n",
ChannelsName(player->channels), TypeName(player->type),
filename);
goto error;
}
/* Allocate enough space for the temp buffer, given the format */
player->datasize = FramesToBytes(BUFFER_SIZE, player->channels,
player->type);
player->data = malloc(player->datasize);
if(player->data == NULL)
{
fprintf(stderr, "Error allocating %d bytes\n", player->datasize);
goto error;
}
return 1;
error:
closeAVFile(player->file);
player->file = NULL;
player->stream = NULL;
player->datasize = 0;
return 0;
}
/* Closes the audio file stream */
static void ClosePlayerFile(StreamPlayer *player)
{
closeAVFile(player->file);
player->file = NULL;
player->stream = NULL;
free(player->data);
player->data = NULL;
player->datasize = 0;
}
/* Prebuffers some audio from the file, and starts playing the source */
static int StartPlayer(StreamPlayer *player)
{
size_t i, got;
/* Rewind the source position and clear the buffer queue */
alSourceRewind(player->source);
alSourcei(player->source, AL_BUFFER, 0);
/* Fill the buffer queue */
for(i = 0;i < NUM_BUFFERS;i++)
{
/* Get some data to give it to the buffer */
got = readAVAudioData(player->stream, player->data, player->datasize);
if(got == 0) break;
palBufferSamplesSOFT(player->buffers[i], player->rate, player->format,
BytesToFrames(got, player->channels, player->type),
player->channels, player->type, player->data);
}
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering for playback\n");
return 0;
}
/* Now queue and start playback! */
alSourceQueueBuffers(player->source, i, player->buffers);
alSourcePlay(player->source);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error starting playback\n");
return 0;
}
return 1;
}
static int UpdatePlayer(StreamPlayer *player)
{
ALint processed, state;
/* Get relevant source info */
alGetSourcei(player->source, AL_SOURCE_STATE, &state);
alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error checking source state\n");
return 0;
}
/* Unqueue and handle each processed buffer */
while(processed > 0)
{
ALuint bufid;
size_t got;
alSourceUnqueueBuffers(player->source, 1, &bufid);
processed--;
/* Read the next chunk of data, refill the buffer, and queue it
* back on the source */
got = readAVAudioData(player->stream, player->data, player->datasize);
if(got > 0)
{
palBufferSamplesSOFT(bufid, player->rate, player->format,
BytesToFrames(got, player->channels, player->type),
player->channels, player->type, player->data);
alSourceQueueBuffers(player->source, 1, &bufid);
}
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering data\n");
return 0;
}
}
/* Make sure the source hasn't underrun */
if(state != AL_PLAYING && state != AL_PAUSED)
{
ALint queued;
/* If no buffers are queued, playback is finished */
alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
if(queued == 0)
return 0;
alSourcePlay(player->source);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error restarting playback\n");
return 0;
}
}
return 1;
}
int main(int argc, char **argv)
{
StreamPlayer *player;
int i;
/* Print out usage if no file was specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s <filenames...>\n", argv[0]);
return 1;
}
if(InitAL() != 0)
return 1;
if(alIsExtensionPresent("AL_SOFT_buffer_samples"))
{
printf("AL_SOFT_buffer_samples supported!\n");
palBufferSamplesSOFT = alGetProcAddress("alBufferSamplesSOFT");
palIsBufferFormatSupportedSOFT = alGetProcAddress("alIsBufferFormatSupportedSOFT");
}
else
printf("AL_SOFT_buffer_samples not supported\n");
player = NewPlayer();
/* Play each file listed on the command line */
for(i = 1;i < argc;i++)
{
if(!OpenPlayerFile(player, argv[i]))
continue;
fprintf(stderr, "Playing %s (%s, %s, %dhz)\n", argv[i],
TypeName(player->type), ChannelsName(player->channels),
player->rate);
if(!StartPlayer(player))
{
ClosePlayerFile(player);
continue;
}
while(UpdatePlayer(player))
Sleep(10);
/* All done with this file. Close it and go to the next */
ClosePlayerFile(player);
}
fprintf(stderr, "Done.\n");
/* All files done. Delete the player, and close OpenAL */
DeletePlayer(player);
player = NULL;
CloseAL();
return 0;
}