diff options
author | Chris Robinson <[email protected]> | 2012-10-14 01:55:39 -0700 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2012-10-14 01:55:39 -0700 |
commit | f7655d44a266a4eeadd7d14921363594b64ac15f (patch) | |
tree | fd89151f62f2b17debe8c9f8f5bc74d24fa35fa6 /examples/common | |
parent | 38e6bfb7024f73671087da2caae5b952b80a5dab (diff) |
Move alhelpers and alffmpeg code to a common sub-directory
Diffstat (limited to 'examples/common')
-rw-r--r-- | examples/common/alffmpeg.c | 638 | ||||
-rw-r--r-- | examples/common/alffmpeg.h | 66 | ||||
-rw-r--r-- | examples/common/alhelpers.c | 319 | ||||
-rw-r--r-- | examples/common/alhelpers.h | 47 |
4 files changed, 1070 insertions, 0 deletions
diff --git a/examples/common/alffmpeg.c b/examples/common/alffmpeg.c new file mode 100644 index 00000000..16f73866 --- /dev/null +++ b/examples/common/alffmpeg.c @@ -0,0 +1,638 @@ +/* + * FFmpeg Decoder Helpers + * + * Copyright (c) 2011 by Chris Robinson <[email protected]> + * + * 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 routines for helping to decode audio using libavformat + * and libavcodec (ffmpeg). There's very little OpenAL-specific code here. */ + +#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" + + +static size_t NextPowerOf2(size_t value) +{ + size_t powerOf2 = 1; + + if(value) + { + value--; + while(value) + { + value >>= 1; + powerOf2 <<= 1; + } + } + return powerOf2; +} + + +struct MemData { + char *buffer; + size_t length; + size_t pos; +}; + +static int MemData_read(void *opaque, uint8_t *buf, int buf_size) +{ + struct MemData *membuf = (struct MemData*)opaque; + int rem = membuf->length - membuf->pos; + + if(rem > buf_size) + rem = buf_size; + + memcpy(buf, &membuf->buffer[membuf->pos], rem); + membuf->pos += rem; + + return rem; +} + +static int MemData_write(void *opaque, uint8_t *buf, int buf_size) +{ + struct MemData *membuf = (struct MemData*)opaque; + int rem = membuf->length - membuf->pos; + + if(rem > buf_size) + rem = buf_size; + + memcpy(&membuf->buffer[membuf->pos], buf, rem); + membuf->pos += rem; + + return rem; +} + +static int64_t MemData_seek(void *opaque, int64_t offset, int whence) +{ + struct MemData *membuf = (struct MemData*)opaque; + + whence &= ~AVSEEK_FORCE; + switch(whence) + { + case SEEK_SET: + if(offset < 0 || (uint64_t)offset > membuf->length) + return -1; + membuf->pos = offset; + break; + + case SEEK_CUR: + if((offset >= 0 && (uint64_t)offset > membuf->length-membuf->pos) || + (offset < 0 && (uint64_t)(-offset) > membuf->pos)) + return -1; + membuf->pos += offset; + break; + + case SEEK_END: + if(offset > 0 || (uint64_t)(-offset) > membuf->length) + return -1; + membuf->pos = membuf->length + offset; + break; + + case AVSEEK_SIZE: + return membuf->length; + + default: + return -1; + } + + return membuf->pos; +} + + +struct PacketList { + AVPacket pkt; + struct PacketList *next; +}; + +struct MyStream { + AVCodecContext *CodecCtx; + int StreamIdx; + + struct PacketList *Packets; + + AVFrame *Frame; + + const uint8_t *FrameData; + size_t FrameDataSize; + + FilePtr parent; +}; + +struct MyFile { + AVFormatContext *FmtCtx; + + StreamPtr *Streams; + size_t StreamsSize; + + struct MemData membuf; +}; + + +static int done_init = 0; + +FilePtr openAVFile(const char *fname) +{ + FilePtr file; + + /* We need to make sure ffmpeg is initialized. Optionally silence warning + * output from the lib */ + if(!done_init) {av_register_all(); + av_log_set_level(AV_LOG_ERROR); + done_init = 1;} + + file = (FilePtr)calloc(1, sizeof(*file)); + if(file && avformat_open_input(&file->FmtCtx, fname, NULL, NULL) == 0) + { + /* After opening, we must search for the stream information because not + * all formats will have it in stream headers */ + if(avformat_find_stream_info(file->FmtCtx, NULL) >= 0) + return file; + avformat_close_input(&file->FmtCtx); + } + + free(file); + return NULL; +} + +FilePtr openAVData(const char *name, char *buffer, size_t buffer_len) +{ + FilePtr file; + + if(!done_init) {av_register_all(); + av_log_set_level(AV_LOG_ERROR); + done_init = 1;} + + if(!name) + name = ""; + + file = (FilePtr)calloc(1, sizeof(*file)); + if(file && (file->FmtCtx=avformat_alloc_context()) != NULL) + { + file->membuf.buffer = buffer; + file->membuf.length = buffer_len; + file->membuf.pos = 0; + + file->FmtCtx->pb = avio_alloc_context(NULL, 0, 0, &file->membuf, + MemData_read, MemData_write, + MemData_seek); + if(file->FmtCtx->pb && avformat_open_input(&file->FmtCtx, name, NULL, NULL) == 0) + { + if(avformat_find_stream_info(file->FmtCtx, NULL) >= 0) + return file; + avformat_close_input(&file->FmtCtx); + } + if(file->FmtCtx) + avformat_free_context(file->FmtCtx); + file->FmtCtx = NULL; + } + + free(file); + return NULL; +} + +FilePtr openAVCustom(const char *name, void *user_data, + int (*read_packet)(void *user_data, uint8_t *buf, int buf_size), + int (*write_packet)(void *user_data, uint8_t *buf, int buf_size), + int64_t (*seek)(void *user_data, int64_t offset, int whence)) +{ + FilePtr file; + + if(!done_init) {av_register_all(); + av_log_set_level(AV_LOG_ERROR); + done_init = 1;} + + if(!name) + name = ""; + + file = (FilePtr)calloc(1, sizeof(*file)); + if(file && (file->FmtCtx=avformat_alloc_context()) != NULL) + { + file->FmtCtx->pb = avio_alloc_context(NULL, 0, 0, user_data, + read_packet, write_packet, seek); + if(file->FmtCtx->pb && avformat_open_input(&file->FmtCtx, name, NULL, NULL) == 0) + { + if(avformat_find_stream_info(file->FmtCtx, NULL) >= 0) + return file; + avformat_close_input(&file->FmtCtx); + } + if(file->FmtCtx) + avformat_free_context(file->FmtCtx); + file->FmtCtx = NULL; + } + + free(file); + return NULL; +} + + +void clearAVAudioData(StreamPtr stream) +{ + while(stream->Packets) + { + struct PacketList *self; + + self = stream->Packets; + stream->Packets = self->next; + + av_free_packet(&self->pkt); + av_free(self); + } +} + + +void closeAVFile(FilePtr file) +{ + size_t i; + + if(!file) return; + + for(i = 0;i < file->StreamsSize;i++) + { + StreamPtr stream = file->Streams[i]; + + while(stream->Packets) + { + struct PacketList *self; + + self = stream->Packets; + stream->Packets = self->next; + + av_free_packet(&self->pkt); + av_free(self); + } + + avcodec_close(stream->CodecCtx); + av_free(stream->Frame); + free(stream); + } + free(file->Streams); + + avformat_close_input(&file->FmtCtx); + free(file); +} + + +int getAVFileInfo(FilePtr file, int *numaudiostreams) +{ + unsigned int i; + int audiocount = 0; + + if(!file) return 1; + for(i = 0;i < file->FmtCtx->nb_streams;i++) + { + if(file->FmtCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) + audiocount++; + } + *numaudiostreams = audiocount; + return 0; +} + +StreamPtr getAVAudioStream(FilePtr file, int streamnum) +{ + unsigned int i; + if(!file) return NULL; + for(i = 0;i < file->FmtCtx->nb_streams;i++) + { + if(file->FmtCtx->streams[i]->codec->codec_type != AVMEDIA_TYPE_AUDIO) + continue; + + if(streamnum == 0) + { + StreamPtr stream; + AVCodec *codec; + void *temp; + size_t j; + + /* Found the requested stream. Check if a handle to this stream + * already exists and return it if it does */ + for(j = 0;j < file->StreamsSize;j++) + { + if(file->Streams[j]->StreamIdx == (int)i) + return file->Streams[j]; + } + + /* Doesn't yet exist. Now allocate a new stream object and fill in + * its info */ + stream = (StreamPtr)calloc(1, sizeof(*stream)); + if(!stream) return NULL; + + stream->parent = file; + stream->CodecCtx = file->FmtCtx->streams[i]->codec; + stream->StreamIdx = i; + + /* Try to find the codec for the given codec ID, and open it */ + codec = avcodec_find_decoder(stream->CodecCtx->codec_id); + if(!codec || avcodec_open2(stream->CodecCtx, codec, NULL) < 0) + { + free(stream); + return NULL; + } + + /* Allocate space for the decoded data to be stored in before it + * gets passed to the app */ + stream->Frame = avcodec_alloc_frame(); + if(!stream->Frame) + { + avcodec_close(stream->CodecCtx); + free(stream); + return NULL; + } + stream->FrameData = NULL; + stream->FrameDataSize = 0; + + /* Append the new stream object to the stream list. The original + * pointer will remain valid if realloc fails, so we need to use + * another pointer to watch for errors and not leak memory */ + temp = realloc(file->Streams, (file->StreamsSize+1) * + sizeof(*file->Streams)); + if(!temp) + { + avcodec_close(stream->CodecCtx); + av_free(stream->Frame); + free(stream); + return NULL; + } + file->Streams = (StreamPtr*)temp; + file->Streams[file->StreamsSize++] = stream; + return stream; + } + streamnum--; + } + return NULL; +} + +int getAVAudioInfo(StreamPtr stream, ALuint *rate, ALenum *channels, ALenum *type) +{ + if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO) + return 1; + + /* Get the sample type for OpenAL given the format detected by ffmpeg. */ + if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_U8) + *type = AL_UNSIGNED_BYTE_SOFT; + else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_S16) + *type = AL_SHORT_SOFT; + else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_S32) + *type = AL_INT_SOFT; + else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_FLT) + *type = AL_FLOAT_SOFT; + else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_DBL) + *type = AL_DOUBLE_SOFT; + else + { + fprintf(stderr, "Unsupported ffmpeg sample format: %s\n", + av_get_sample_fmt_name(stream->CodecCtx->sample_fmt)); + return 1; + } + + /* Get the OpenAL channel configuration using the channel layout detected + * by ffmpeg. NOTE: some file types may not specify a channel layout. In + * that case, one must be guessed based on the channel count. */ + if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_MONO) + *channels = AL_MONO_SOFT; + else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_STEREO) + *channels = AL_STEREO_SOFT; + else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_QUAD) + *channels = AL_QUAD_SOFT; + else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) + *channels = AL_5POINT1_SOFT; + else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1) + *channels = AL_7POINT1_SOFT; + else if(stream->CodecCtx->channel_layout == 0) + { + /* Unknown channel layout. Try to guess. */ + if(stream->CodecCtx->channels == 1) + *channels = AL_MONO_SOFT; + else if(stream->CodecCtx->channels == 2) + *channels = AL_STEREO_SOFT; + else + { + fprintf(stderr, "Unsupported ffmpeg raw channel count: %d\n", + stream->CodecCtx->channels); + return 1; + } + } + else + { + char str[1024]; + av_get_channel_layout_string(str, sizeof(str), stream->CodecCtx->channels, + stream->CodecCtx->channel_layout); + fprintf(stderr, "Unsupported ffmpeg channel layout: %s\n", str); + return 1; + } + + *rate = stream->CodecCtx->sample_rate; + + return 0; +} + + +/* Used by getAV*Data to search for more compressed data, and buffer it in the + * correct stream. It won't buffer data for streams that the app doesn't have a + * handle for. */ +static int getNextPacket(FilePtr file, int streamidx) +{ + struct PacketList *packet; + + packet = (struct PacketList*)av_malloc(sizeof(*packet)); + packet->next = NULL; + +next_packet: + while(av_read_frame(file->FmtCtx, &packet->pkt) >= 0) + { + StreamPtr *iter = file->Streams; + StreamPtr *iter_end = iter + file->StreamsSize; + + /* Check each stream the user has a handle for, looking for the one + * this packet belongs to */ + while(iter != iter_end) + { + if((*iter)->StreamIdx == packet->pkt.stream_index) + { + struct PacketList **last; + + last = &(*iter)->Packets; + while(*last != NULL) + last = &(*last)->next; + + *last = packet; + if((*iter)->StreamIdx == streamidx) + return 1; + + packet = (struct PacketList*)av_malloc(sizeof(*packet)); + packet->next = NULL; + goto next_packet; + } + iter++; + } + /* Free the packet and look for another */ + av_free_packet(&packet->pkt); + } + + av_free(packet); + return 0; +} + +uint8_t *getAVAudioData(StreamPtr stream, size_t *length) +{ + int got_frame; + int len; + + if(length) *length = 0; + + if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO) + return NULL; + +next_packet: + if(!stream->Packets && !getNextPacket(stream->parent, stream->StreamIdx)) + return NULL; + + /* Decode some data, and check for errors */ + avcodec_get_frame_defaults(stream->Frame); + while((len=avcodec_decode_audio4(stream->CodecCtx, stream->Frame, + &got_frame, &stream->Packets->pkt)) < 0) + { + struct PacketList *self; + + /* Error? Drop it and try the next, I guess... */ + self = stream->Packets; + stream->Packets = self->next; + + av_free_packet(&self->pkt); + av_free(self); + + if(!stream->Packets) + goto next_packet; + } + + if(len < stream->Packets->pkt.size) + { + /* Move the unread data to the front and clear the end bits */ + int remaining = stream->Packets->pkt.size - len; + memmove(stream->Packets->pkt.data, &stream->Packets->pkt.data[len], + remaining); + memset(&stream->Packets->pkt.data[remaining], 0, + stream->Packets->pkt.size - remaining); + stream->Packets->pkt.size -= len; + } + else + { + struct PacketList *self; + + self = stream->Packets; + stream->Packets = self->next; + + av_free_packet(&self->pkt); + av_free(self); + } + + if(!got_frame || stream->Frame->nb_samples == 0) + goto next_packet; + + /* Set the output buffer size */ + *length = av_samples_get_buffer_size(NULL, stream->CodecCtx->channels, + stream->Frame->nb_samples, + stream->CodecCtx->sample_fmt, 1); + + return stream->Frame->data[0]; +} + +size_t readAVAudioData(StreamPtr stream, void *data, size_t length) +{ + size_t dec = 0; + + if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO) + return 0; + + while(dec < length) + { + /* If there's no decoded data, find some */ + if(stream->FrameDataSize == 0) + { + stream->FrameData = getAVAudioData(stream, &stream->FrameDataSize); + if(!stream->FrameData) + break; + } + + if(stream->FrameDataSize > 0) + { + /* Get the amount of bytes remaining to be written, and clamp to + * the amount of decoded data we have */ + size_t rem = length-dec; + if(rem > stream->FrameDataSize) + rem = stream->FrameDataSize; + + /* Copy the data to the app's buffer and increment */ + if(data != NULL) + { + memcpy(data, stream->FrameData, rem); + data = (char*)data + rem; + } + dec += rem; + + /* If there's any decoded data left, move it to the front of the + * buffer for next time */ + stream->FrameData += rem; + stream->FrameDataSize -= rem; + } + } + + /* Return the number of bytes we were able to get */ + return dec; +} + +void *decodeAVAudioStream(StreamPtr stream, size_t *length) +{ + char *outbuf = NULL; + size_t buflen = 0; + void *inbuf; + size_t got; + + *length = 0; + if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO) + return NULL; + + while((inbuf=getAVAudioData(stream, &got)) != NULL && got > 0) + { + void *ptr; + + ptr = realloc(outbuf, NextPowerOf2(buflen+got)); + if(ptr == NULL) + break; + outbuf = (char*)ptr; + + memcpy(&outbuf[buflen], inbuf, got); + buflen += got; + } + outbuf = (char*)realloc(outbuf, buflen); + + *length = buflen; + return outbuf; +} diff --git a/examples/common/alffmpeg.h b/examples/common/alffmpeg.h new file mode 100644 index 00000000..27e49ab4 --- /dev/null +++ b/examples/common/alffmpeg.h @@ -0,0 +1,66 @@ +#ifndef ALFFMPEG_H +#define ALFFMPEG_H + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> + +/* Opaque handles to files and streams. Apps don't need to concern themselves + * with the internals */ +typedef struct MyFile *FilePtr; +typedef struct MyStream *StreamPtr; + +/* Opens a file with ffmpeg and sets up the streams' information */ +FilePtr openAVFile(const char *fname); + +/* Opens a named file image with ffmpeg and sets up the streams' information */ +FilePtr openAVData(const char *name, char *buffer, size_t buffer_len); + +/* Opens a named data stream with ffmpeg, using the specified data pointer and + * callbacks, and sets up the streams' information */ +FilePtr openAVCustom(const char *name, void *user_data, + int (*read_packet)(void *user_data, uint8_t *buf, int buf_size), + int (*write_packet)(void *user_data, uint8_t *buf, int buf_size), + int64_t (*seek)(void *user_data, int64_t offset, int whence)); + +/* Closes/frees an opened file and any of its streams */ +void closeAVFile(FilePtr file); + +/* Reports certain information from the file, eg, the number of audio + * streams. Returns 0 on success. */ +int getAVFileInfo(FilePtr file, int *numaudiostreams); + +/* Retrieves a handle for the given audio stream number (generally 0, but some + * files can have multiple audio streams in one file). */ +StreamPtr getAVAudioStream(FilePtr file, int streamnum); + +/* Returns information about the given audio stream. Returns 0 on success. */ +int getAVAudioInfo(StreamPtr stream, ALuint *rate, ALenum *channels, ALenum *type); + +/* Returns a pointer to the next available packet of decoded audio. Any data + * from a previously-decoded packet is dropped. The size (in bytes) of the + * returned data buffer is stored in 'length', and the returned pointer is only + * valid until the next call to getAVAudioData or readAVAudioData. */ +uint8_t *getAVAudioData(StreamPtr stream, size_t *length); + +/* The "meat" function. Decodes audio and writes, at most, length bytes into + * the provided data buffer. Will only return less for end-of-stream or error + * conditions. Returns the number of bytes written. */ +size_t readAVAudioData(StreamPtr stream, void *data, size_t length); + +/* Decodes all remaining data from the stream and returns a buffer containing + * the audio data, with the size stored in 'length'. The returned pointer must + * be freed with a call to free(). Note that since this decodes the whole + * stream, using it on lengthy streams (eg, music) will use a lot of memory. + * Such streams are better handled using getAVAudioData or readAVAudioData to + * keep smaller chunks in memory at any given time. */ +void *decodeAVAudioStream(StreamPtr stream, size_t *length); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif /* ALFFMPEG_H */ diff --git a/examples/common/alhelpers.c b/examples/common/alhelpers.c new file mode 100644 index 00000000..dbfe2383 --- /dev/null +++ b/examples/common/alhelpers.c @@ -0,0 +1,319 @@ +/* + * OpenAL Helpers + * + * Copyright (c) 2011 by Chris Robinson <[email protected]> + * + * 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 routines to help with some menial OpenAL-related tasks, + * such as opening a device and setting up a context, closing the device and + * destroying its context, converting between frame counts and byte lengths, + * finding an appropriate buffer format, and getting readable strings for + * channel configs and sample types. */ + +#include <stdio.h> + +#include "AL/al.h" +#include "AL/alc.h" +#include "AL/alext.h" + +#include "alhelpers.h" + + +const char *ChannelsName(ALenum chans) +{ + switch(chans) + { + case AL_MONO_SOFT: return "Mono"; + case AL_STEREO_SOFT: return "Stereo"; + case AL_REAR_SOFT: return "Rear"; + case AL_QUAD_SOFT: return "Quadraphonic"; + case AL_5POINT1_SOFT: return "5.1 Surround"; + case AL_6POINT1_SOFT: return "6.1 Surround"; + case AL_7POINT1_SOFT: return "7.1 Surround"; + } + return "Unknown Channels"; +} + +const char *TypeName(ALenum type) +{ + switch(type) + { + case AL_BYTE_SOFT: return "S8"; + case AL_UNSIGNED_BYTE_SOFT: return "U8"; + case AL_SHORT_SOFT: return "S16"; + case AL_UNSIGNED_SHORT_SOFT: return "U16"; + case AL_INT_SOFT: return "S32"; + case AL_UNSIGNED_INT_SOFT: return "U32"; + case AL_FLOAT_SOFT: return "Float32"; + case AL_DOUBLE_SOFT: return "Float64"; + } + return "Unknown Type"; +} + + +ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type) +{ + switch(channels) + { + case AL_MONO_SOFT: size *= 1; break; + case AL_STEREO_SOFT: size *= 2; break; + case AL_REAR_SOFT: size *= 2; break; + case AL_QUAD_SOFT: size *= 4; break; + case AL_5POINT1_SOFT: size *= 6; break; + case AL_6POINT1_SOFT: size *= 7; break; + case AL_7POINT1_SOFT: size *= 8; break; + } + + switch(type) + { + case AL_BYTE_SOFT: size *= sizeof(ALbyte); break; + case AL_UNSIGNED_BYTE_SOFT: size *= sizeof(ALubyte); break; + case AL_SHORT_SOFT: size *= sizeof(ALshort); break; + case AL_UNSIGNED_SHORT_SOFT: size *= sizeof(ALushort); break; + case AL_INT_SOFT: size *= sizeof(ALint); break; + case AL_UNSIGNED_INT_SOFT: size *= sizeof(ALuint); break; + case AL_FLOAT_SOFT: size *= sizeof(ALfloat); break; + case AL_DOUBLE_SOFT: size *= sizeof(ALdouble); break; + } + + return size; +} + +ALsizei BytesToFrames(ALsizei size, ALenum channels, ALenum type) +{ + return size / FramesToBytes(1, channels, type); +} + + +ALenum GetFormat(ALenum channels, ALenum type, LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT) +{ + ALenum format = AL_NONE; + + /* If using AL_SOFT_buffer_samples, try looking through its formats */ + if(palIsBufferFormatSupportedSOFT) + { + /* AL_SOFT_buffer_samples is more lenient with matching formats. The + * specified sample type does not need to match the returned format, + * but it is nice to try to get something close. */ + if(type == AL_UNSIGNED_BYTE_SOFT || type == AL_BYTE_SOFT) + { + if(channels == AL_MONO_SOFT) format = AL_MONO8_SOFT; + else if(channels == AL_STEREO_SOFT) format = AL_STEREO8_SOFT; + else if(channels == AL_QUAD_SOFT) format = AL_QUAD8_SOFT; + else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_8_SOFT; + else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_8_SOFT; + else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_8_SOFT; + } + else if(type == AL_UNSIGNED_SHORT_SOFT || type == AL_SHORT_SOFT) + { + if(channels == AL_MONO_SOFT) format = AL_MONO16_SOFT; + else if(channels == AL_STEREO_SOFT) format = AL_STEREO16_SOFT; + else if(channels == AL_QUAD_SOFT) format = AL_QUAD16_SOFT; + else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_16_SOFT; + else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_16_SOFT; + else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_16_SOFT; + } + else if(type == AL_UNSIGNED_BYTE3_SOFT || type == AL_BYTE3_SOFT || + type == AL_UNSIGNED_INT_SOFT || type == AL_INT_SOFT || + type == AL_FLOAT_SOFT || type == AL_DOUBLE_SOFT) + { + if(channels == AL_MONO_SOFT) format = AL_MONO32F_SOFT; + else if(channels == AL_STEREO_SOFT) format = AL_STEREO32F_SOFT; + else if(channels == AL_QUAD_SOFT) format = AL_QUAD32F_SOFT; + else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_32F_SOFT; + else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_32F_SOFT; + else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_32F_SOFT; + } + + if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) + format = AL_NONE; + + /* A matching format was not found or supported. Try 32-bit float. */ + if(format == AL_NONE) + { + if(channels == AL_MONO_SOFT) format = AL_MONO32F_SOFT; + else if(channels == AL_STEREO_SOFT) format = AL_STEREO32F_SOFT; + else if(channels == AL_QUAD_SOFT) format = AL_QUAD32F_SOFT; + else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_32F_SOFT; + else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_32F_SOFT; + else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_32F_SOFT; + + if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) + format = AL_NONE; + } + /* 32-bit float not supported. Try 16-bit int. */ + if(format == AL_NONE) + { + if(channels == AL_MONO_SOFT) format = AL_MONO16_SOFT; + else if(channels == AL_STEREO_SOFT) format = AL_STEREO16_SOFT; + else if(channels == AL_QUAD_SOFT) format = AL_QUAD16_SOFT; + else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_16_SOFT; + else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_16_SOFT; + else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_16_SOFT; + + if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) + format = AL_NONE; + } + /* 16-bit int not supported. Try 8-bit int. */ + if(format == AL_NONE) + { + if(channels == AL_MONO_SOFT) format = AL_MONO8_SOFT; + else if(channels == AL_STEREO_SOFT) format = AL_STEREO8_SOFT; + else if(channels == AL_QUAD_SOFT) format = AL_QUAD8_SOFT; + else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_8_SOFT; + else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_8_SOFT; + else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_8_SOFT; + + if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format)) + format = AL_NONE; + } + + return format; + } + + /* We use the AL_EXT_MCFORMATS extension to provide output of Quad, 5.1, + * and 7.1 channel configs, AL_EXT_FLOAT32 for 32-bit float samples, and + * AL_EXT_DOUBLE for 64-bit float samples. */ + if(type == AL_UNSIGNED_BYTE_SOFT) + { + if(channels == AL_MONO_SOFT) + format = AL_FORMAT_MONO8; + else if(channels == AL_STEREO_SOFT) + format = AL_FORMAT_STEREO8; + else if(alIsExtensionPresent("AL_EXT_MCFORMATS")) + { + if(channels == AL_QUAD_SOFT) + format = alGetEnumValue("AL_FORMAT_QUAD8"); + else if(channels == AL_5POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_51CHN8"); + else if(channels == AL_6POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_61CHN8"); + else if(channels == AL_7POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_71CHN8"); + } + } + else if(type == AL_SHORT_SOFT) + { + if(channels == AL_MONO_SOFT) + format = AL_FORMAT_MONO16; + else if(channels == AL_STEREO_SOFT) + format = AL_FORMAT_STEREO16; + else if(alIsExtensionPresent("AL_EXT_MCFORMATS")) + { + if(channels == AL_QUAD_SOFT) + format = alGetEnumValue("AL_FORMAT_QUAD16"); + else if(channels == AL_5POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_51CHN16"); + else if(channels == AL_6POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_61CHN16"); + else if(channels == AL_7POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_71CHN16"); + } + } + else if(type == AL_FLOAT_SOFT && alIsExtensionPresent("AL_EXT_FLOAT32")) + { + if(channels == AL_MONO_SOFT) + format = alGetEnumValue("AL_FORMAT_MONO_FLOAT32"); + else if(channels == AL_STEREO_SOFT) + format = alGetEnumValue("AL_FORMAT_STEREO_FLOAT32"); + else if(alIsExtensionPresent("AL_EXT_MCFORMATS")) + { + if(channels == AL_QUAD_SOFT) + format = alGetEnumValue("AL_FORMAT_QUAD32"); + else if(channels == AL_5POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_51CHN32"); + else if(channels == AL_6POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_61CHN32"); + else if(channels == AL_7POINT1_SOFT) + format = alGetEnumValue("AL_FORMAT_71CHN32"); + } + } + else if(type == AL_DOUBLE_SOFT && alIsExtensionPresent("AL_EXT_DOUBLE")) + { + if(channels == AL_MONO_SOFT) + format = alGetEnumValue("AL_FORMAT_MONO_DOUBLE"); + else if(channels == AL_STEREO_SOFT) + format = alGetEnumValue("AL_FORMAT_STEREO_DOUBLE"); + } + + /* NOTE: It seems OSX returns -1 from alGetEnumValue for unknown enums, as + * opposed to 0. Correct it. */ + if(format == -1) + format = 0; + + return format; +} + + +void AL_APIENTRY wrap_BufferSamples(ALuint buffer, ALuint samplerate, + ALenum internalformat, ALsizei samples, + ALenum channels, ALenum type, + const ALvoid *data) +{ + alBufferData(buffer, internalformat, data, + FramesToBytes(samples, channels, type), + samplerate); +} + + +int InitAL(void) +{ + ALCdevice *device; + ALCcontext *ctx; + + /* Open and initialize a device with default settings */ + device = alcOpenDevice(NULL); + if(!device) + { + fprintf(stderr, "Could not open a device!\n"); + return 1; + } + + ctx = alcCreateContext(device, NULL); + if(ctx == NULL || alcMakeContextCurrent(ctx) == ALC_FALSE) + { + if(ctx != NULL) + alcDestroyContext(ctx); + alcCloseDevice(device); + fprintf(stderr, "Could not set a context!\n"); + return 1; + } + + return 0; +} + +void CloseAL(void) +{ + ALCdevice *device; + ALCcontext *ctx; + + /* Close the device belonging to the current context, and destroy the + * context. */ + ctx = alcGetCurrentContext(); + if(ctx == NULL) + return; + + device = alcGetContextsDevice(ctx); + + alcMakeContextCurrent(NULL); + alcDestroyContext(ctx); + alcCloseDevice(device); +} diff --git a/examples/common/alhelpers.h b/examples/common/alhelpers.h new file mode 100644 index 00000000..eda8925e --- /dev/null +++ b/examples/common/alhelpers.h @@ -0,0 +1,47 @@ +#ifndef ALHELPERS_H +#define ALHELPERS_H + +#ifndef _WIN32 +#include <unistd.h> +#define Sleep(x) usleep((x)*1000) +#else +#define WIN32_LEAN_AND_MEAN +#include <windows.h> +#endif + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + +/* Some helper functions to get the name from the channel and type enums. */ +const char *ChannelsName(ALenum chans); +const char *TypeName(ALenum type); + +/* Helpers to convert frame counts and byte lengths. */ +ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type); +ALsizei BytesToFrames(ALsizei size, ALenum channels, ALenum type); + +/* Retrieves a compatible buffer format given the channel configuration and + * sample type. If an alIsBufferFormatSupportedSOFT-compatible function is + * provided, it will be called to find the closest-matching format from + * AL_SOFT_buffer_samples. Returns AL_NONE (0) if no supported format can be + * found. */ +ALenum GetFormat(ALenum channels, ALenum type, LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT); + +/* Loads samples into a buffer using the standard alBufferData call, but with a + * LPALBUFFERSAMPLESSOFT-compatible prototype. Assumes internalformat is valid + * for alBufferData, and that channels and type match it. */ +void AL_APIENTRY wrap_BufferSamples(ALuint buffer, ALuint samplerate, + ALenum internalformat, ALsizei samples, + ALenum channels, ALenum type, + const ALvoid *data); + +/* Easy device init/deinit functions. InitAL returns 0 on success. */ +int InitAL(void); +void CloseAL(void); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif /* ALHELPERS_H */ |