fteqw/plugins/avplug/avaudio.c
Spoike 4c2066601a Support connecting subnodes to servers over tcp (instead of depending on fork).
Fixed up the -netquake / -spasm / -fitz args slightly, should actually be usable now.
sv_mintic 0 is now treated as 0.013 when using nqplayerphysics, to try to make it smoother for nq clients.
Preparing for astc's volume formats. Mostly for completeness, I was bored. Disabled for now because nothing supports them anyway.
Fix broken mousewheel in SDL2 builds.
Fix configs not getting loaded following initial downloads in the web port/etc.
Make the near-cloud layer of q1 scrolling sky fully opaque by default (like vanilla).
Sky fog now ignores depth, treating it as an infinite distance.
Fix turbs not responding to fog.
r_fullbright no longer needs vid_reload to take effect (and more efficient now).
Tweaked the audio code to use an format enum instead of byte width, just with the same values still, primarily to clean up loaders that deal with S32 vs F32, or U8 vs S8.
Added a cvar to control whether to use threads for the qcgc. Still disabled by default but no longer requires engine recompiles to enable!



git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5683 fc73d0e0-1445-4013-8a0c-d673dee63da5
2020-04-29 10:43:22 +00:00

557 lines
15 KiB
C

#include "../plugin.h"
#include "../engine.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
static size_t activedecoders;
static cvar_t *ffmpeg_audiodecoder, *pdeveloper;
#define HAVE_DECOUPLED_API (LIBAVCODEC_VERSION_MAJOR>57 || (LIBAVCODEC_VERSION_MAJOR==57&&LIBAVCODEC_VERSION_MINOR>=36))
struct avaudioctx
{
//raw file
uint8_t *filedata;
size_t fileofs;
size_t filesize;
//avformat stuff
AVFormatContext *pFormatCtx;
int audioStream;
AVCodecContext *pACodecCtx;
AVFrame *pAFrame;
//decoding
int64_t lasttime;
//output audio
//we throw away data if the format changes. which is awkward, but gah.
int64_t samples_framestart;
int samples_channels;
int samples_speed;
qaudiofmt_t samples_format;
qbyte *samples_buffer;
size_t samples_framecount;
size_t samples_maxbytes;
};
static void S_AV_Purge(sfx_t *s)
{
struct avaudioctx *ctx = (struct avaudioctx*)s->decoder.buf;
s->loadstate = SLS_NOTLOADED;
// Free the audio decoder
if (ctx->pACodecCtx)
avcodec_close(ctx->pACodecCtx);
av_free(ctx->pAFrame);
// Close the video file
avformat_close_input(&ctx->pFormatCtx);
//free the decoded buffer
free(ctx->samples_buffer);
//file storage will be cleared here too
free(ctx);
if (s->decoder.ended)
activedecoders--;
memset(&s->decoder, 0, sizeof(s->decoder));
}
#define QAF_U8 0x81
#define QAF_S32 0x04
#ifndef MIXER_F32
#define QAF_F32 0x84
#endif
#define QAF_F64 0x88
static void S_AV_ReadFrame(struct avaudioctx *ctx)
{ //reads an audioframe and spits its data into the output sound file for the game engine to use.
qaudiofmt_t outformat = QAF_S16, informat=QAF_S16;
int channels = ctx->pACodecCtx->channels;
int planes = 1, p;
unsigned int auddatasize = av_samples_get_buffer_size(NULL, ctx->pACodecCtx->channels, ctx->pAFrame->nb_samples, ctx->pACodecCtx->sample_fmt, 1);
switch(ctx->pACodecCtx->sample_fmt)
{ //we don't support planar audio. we just treat it as mono instead.
default:
auddatasize = 0;
break;
case AV_SAMPLE_FMT_U8P:
planes = channels;
outformat = QAF_S8;
informat = QAF_U8;
break;
case AV_SAMPLE_FMT_U8:
planes = 1;
outformat = QAF_S8;
informat = QAF_U8;
break;
case AV_SAMPLE_FMT_S16P:
planes = channels;
outformat = QAF_S16;
informat = QAF_S16;
break;
case AV_SAMPLE_FMT_S16:
planes = 1;
outformat = QAF_S16;
informat = QAF_S16;
break;
case AV_SAMPLE_FMT_S32P:
planes = channels;
outformat = QAF_S16;
informat = QAF_S32;
break;
case AV_SAMPLE_FMT_S32:
planes = 1;
outformat = QAF_S16;
informat = QAF_S32;
break;
#ifdef MIXER_F32
case AV_SAMPLE_FMT_FLTP:
planes = channels;
outformat = QAF_F32;
informat = QAF_F32;
break;
case AV_SAMPLE_FMT_FLT:
planes = 1;
outformat = QAF_F32;
informat = QAF_F32;
break;
case AV_SAMPLE_FMT_DBLP:
planes = channels;
outformat = QAF_F32;
informat = QAF_F64;
break;
case AV_SAMPLE_FMT_DBL:
planes = 1;
outformat = QAF_F32;
informat = QAF_F64;
break;
#else
case AV_SAMPLE_FMT_FLTP:
planes = channels;
outformat = QAF_S16;
informat = QAF_F32;
break;
case AV_SAMPLE_FMT_FLT:
planes = 1;
outformat = QAF_S16;
informat = QAF_F32;
break;
case AV_SAMPLE_FMT_DBLP:
planes = channels;
outformat = QAF_S16;
informat = QAF_F64;
break;
case AV_SAMPLE_FMT_DBL:
planes = 1;
outformat = QAF_S16;
informat = QAF_F64;
break;
#endif
}
if (ctx->samples_channels != channels || ctx->samples_speed != ctx->pACodecCtx->sample_rate || ctx->samples_format != outformat)
{ //something changed, update
ctx->samples_channels = channels;
ctx->samples_speed = ctx->pACodecCtx->sample_rate;
ctx->samples_format = outformat;
//and discard any decoded audio. this might loose some.
ctx->samples_framestart += ctx->samples_framecount;
ctx->samples_framecount = 0;
}
if (ctx->samples_maxbytes < (ctx->samples_framecount*QAF_BYTES(ctx->samples_format)*ctx->samples_channels)+auddatasize)
{
ctx->samples_maxbytes = (ctx->samples_framecount*QAF_BYTES(ctx->samples_format)*ctx->samples_channels)+auddatasize;
ctx->samples_maxbytes *= 2; //slop
ctx->samples_buffer = realloc(ctx->samples_buffer, ctx->samples_maxbytes);
}
if (planes==1 && outformat != QAF_S8 && informat==outformat)
memcpy(ctx->samples_buffer + ctx->samples_framecount*(QAF_BYTES(ctx->samples_format)*ctx->samples_channels), ctx->pAFrame->data[0], auddatasize);
else
{
void *fte_restrict outv = (ctx->samples_buffer + ctx->samples_framecount*(QAF_BYTES(ctx->samples_format)*ctx->samples_channels));
size_t i, samples = auddatasize / (planes*QAF_BYTES(informat));
if (outformat == QAF_S8 && informat == QAF_U8)
{
char *out = outv;
for (p = 0; p < planes; p++, out++)
{
unsigned char *in = ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
out[i*planes] = in[i]-128; //convert from u8 to s8.
}
}
else if (outformat == QAF_S16 && informat == QAF_S16)
{
signed short *out = outv;
for (p = 0; p < planes; p++, out++)
{
signed short *in = (signed short *)ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
out[i*planes] = in[i]; //no conversion needed
}
}
else if (outformat == QAF_S16 && informat == QAF_S32)
{
signed short *out = outv;
for (p = 0; p < planes; p++, out++)
{
signed int *in = (signed int *)ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
out[i*planes] = in[i]>>16; //just use the MSBs, no clamping needed.
}
}
#ifdef MIXER_F32
else if (outformat == QAF_F32 && informat == QAF_F32)
{
float *out = outv;
for (p = 0; p < planes; p++, out++)
{
float *in = (float *)ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
out[i*planes] = in[i]; //no conversion needed.
}
}
else if (outformat == QAF_F32 && informat == QAF_F64)
{
float *out = outv;
for (p = 0; p < planes; p++, out++)
{
double *in = (double *)ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
out[i*planes] = in[i]; //no clamping needed.
}
}
#else
else if (outformat == QAF_S16 && informat == QAF_F32)
{
signed short *out = outv;
for (p = 0; p < planes; p++, out++)
{
float *in = (float *)ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
{
int v = in[i] * 32767;
if (v < -32768)
v = -32768;
if (v > 32767)
v = 32767;
out[i*planes] = v;
}
}
}
else if (outformat == QAF_S16 && informat == QAF_F64)
{
signed short *out = outv;
for (p = 0; p < planes; p++, out++)
{
double *in = (double *)ctx->pAFrame->data[p];
for (i = 0; i < samples; i++)
{
int v = in[i] * 32767;
if (v < -32768)
v = -32768;
if (v > 32767)
v = 32767;
out[i*planes] = v;
}
}
}
#endif
}
ctx->samples_framecount += auddatasize/(QAF_BYTES(informat)*ctx->samples_channels);
}
static sfxcache_t *S_AV_Locate(sfx_t *sfx, sfxcache_t *buf, ssamplepos_t start, int length)
{ //warning: can be called on a different thread.
struct avaudioctx *ctx = (struct avaudioctx*)sfx->decoder.buf;
AVPacket packet;
int64_t curtime;
if (!buf)
return NULL;
curtime = start + length;
while (1)
{
if (start < ctx->samples_framestart)
break; //o.O rewind!
if (ctx->samples_framestart+ctx->samples_framecount > curtime)
break; //no need yet.
#ifdef HAVE_DECOUPLED_API
if(0==avcodec_receive_frame(ctx->pACodecCtx, ctx->pAFrame))
{
S_AV_ReadFrame(ctx);
continue;
}
#endif
// We're ahead of the previous frame. try and read the next.
if (av_read_frame(ctx->pFormatCtx, &packet) < 0)
break;
// Is this a packet from the video stream?
if(packet.stream_index==ctx->audioStream)
{
#ifdef HAVE_DECOUPLED_API
avcodec_send_packet(ctx->pACodecCtx, &packet);
#else
int okay;
int len;
void *odata = packet.data;
while (packet.size > 0)
{ //this old api only decodes part of the packet with each itteration, so keep reading until we decoded the entire thing.
okay = false;
len = avcodec_decode_audio4(ctx->pACodecCtx, ctx->pAFrame, &okay, &packet);
if (len < 0)
break;
packet.size -= len;
packet.data += len;
if (okay)
S_AV_ReadFrame(ctx);
}
packet.data = odata;
#endif
}
// Free the packet that was allocated by av_read_frame
av_packet_unref(&packet);
}
buf->length = ctx->samples_framecount;
buf->speed = ctx->samples_speed;
buf->format = ctx->samples_format;
buf->numchannels = ctx->samples_channels;
buf->soundoffset = ctx->samples_framestart;
buf->data = ctx->samples_buffer;
//if we couldn't return any new data, then we're at an eof, return NULL to signal that.
if (start == buf->soundoffset + buf->length && length > 0)
return NULL;
return buf;
}
static float S_AV_Query(struct sfx_s *sfx, struct sfxcache_s *buf, char *title, size_t titlesize)
{
struct avaudioctx *ctx = (struct avaudioctx*)sfx->decoder.buf;
if (!ctx)
return -1;
if (buf)
{
buf->data = NULL;
buf->soundoffset = 0;
buf->length = 0;
buf->numchannels = ctx->samples_channels;
buf->speed = ctx->samples_speed;
buf->format = ctx->samples_format;
}
return ctx->pFormatCtx->duration / (float)AV_TIME_BASE;
}
static int AVIO_Mem_Read(void *opaque, uint8_t *buf, int buf_size)
{
struct avaudioctx *ctx = opaque;
if (ctx->fileofs > ctx->filesize)
buf_size = 0;
if (buf_size > ctx->filesize-ctx->fileofs)
buf_size = ctx->filesize-ctx->fileofs;
if (buf_size > 0)
{
memcpy(buf, ctx->filedata + ctx->fileofs, buf_size);
ctx->fileofs += buf_size;
return buf_size;
}
return 0;
}
static int64_t AVIO_Mem_Seek(void *opaque, int64_t offset, int whence)
{
struct avaudioctx *ctx = opaque;
whence &= ~AVSEEK_FORCE;
switch(whence)
{
default:
return -1;
case SEEK_SET:
ctx->fileofs = offset;
break;
case SEEK_CUR:
ctx->fileofs += offset;
break;
case SEEK_END:
ctx->fileofs = ctx->filesize + offset;
break;
case AVSEEK_SIZE:
return ctx->filesize;
}
if (ctx->fileofs < 0)
ctx->fileofs = 0;
return ctx->fileofs;
}
/*const char *COM_GetFileExtension (const char *in)
{
const char *dot;
for (dot = in + strlen(in); dot >= in && *dot != '.'; dot--)
;
if (dot < in)
return "";
in = dot+1;
return in;
}*/
static qboolean QDECL S_LoadAVSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed)
{
struct avaudioctx *ctx;
int i;
AVCodec *pCodec;
const int iBufSize = 4 * 1024;
if (!ffmpeg_audiodecoder)
return false;
if (!ffmpeg_audiodecoder->ival /* && *ffmpeg_audiodecoder.string */)
return false;
if (!data || !datalen)
return false;
//ignore it if it looks like a wav file. that means we don't need to figure out how to calculate loopstart.
//FIXME: this also blocks playing the audio from avi files too!
if (datalen >= 4 && !strncmp(data, "RIFF", 4))
return false;
// if (strcasecmp(COM_GetFileExtension(s->name), "wav")) //don't do .wav - I've no idea how to read the loopstart tag with ffmpeg.
// return false;
s->decoder.buf = ctx = malloc(sizeof(*ctx) + datalen);
if (!ctx)
return false; //o.O
memset(ctx, 0, sizeof(*ctx));
// Create internal io buffer for FFmpeg
ctx->filedata = data; //defer that copy
ctx->filesize = datalen; //defer that copy
ctx->pFormatCtx = avformat_alloc_context();
ctx->pFormatCtx->pb = avio_alloc_context(av_malloc(iBufSize), iBufSize, 0, ctx, AVIO_Mem_Read, 0, AVIO_Mem_Seek);
// Open file
if(avformat_open_input(&ctx->pFormatCtx, s->name, NULL, NULL)==0)
{
// Retrieve stream information
if(avformat_find_stream_info(ctx->pFormatCtx, NULL)>=0)
{
ctx->audioStream=-1;
for(i=0; i<ctx->pFormatCtx->nb_streams; i++)
#if LIBAVFORMAT_VERSION_MAJOR >= 57
if(ctx->pFormatCtx->streams[i]->codecpar->codec_type==AVMEDIA_TYPE_AUDIO)
#else
if(ctx->pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO)
#endif
{
ctx->audioStream=i;
break;
}
if(ctx->audioStream!=-1)
{
#if LIBAVFORMAT_VERSION_MAJOR >= 57
pCodec=avcodec_find_decoder(ctx->pFormatCtx->streams[ctx->audioStream]->codecpar->codec_id);
ctx->pACodecCtx = avcodec_alloc_context3(pCodec);
if (avcodec_parameters_to_context(ctx->pACodecCtx, ctx->pFormatCtx->streams[ctx->audioStream]->codecpar) < 0)
{
avcodec_free_context(&ctx->pACodecCtx);
pCodec = NULL;
}
#else
ctx->pACodecCtx=ctx->pFormatCtx->streams[ctx->audioStream]->codec;
pCodec=avcodec_find_decoder(ctx->pACodecCtx->codec_id);
#endif
ctx->pAFrame=av_frame_alloc();
if(pCodec!=NULL && ctx->pAFrame && avcodec_open2(ctx->pACodecCtx, pCodec, NULL) >= 0)
{ //success
}
else
ctx->audioStream = -1;
}
}
if (ctx->audioStream != -1)
{
//sucky copy
ctx->filedata = (uint8_t*)(ctx+1);
memcpy(ctx->filedata, data, datalen);
s->decoder.ended = S_AV_Purge;
s->decoder.purge = S_AV_Purge;
s->decoder.decodedata = S_AV_Locate;
s->decoder.querydata = S_AV_Query;
activedecoders++;
return true;
}
}
S_AV_Purge(s);
return false;
}
qboolean AVAudio_MayUnload(void)
{
return activedecoders==0;
}
static qboolean AVAudio_Init(void)
{
if (!plugfuncs->ExportFunction("MayUnload", AVAudio_MayUnload) ||
!plugfuncs->ExportFunction("S_LoadSound", S_LoadAVSound))
{
Con_Printf("ffmpeg: Engine doesn't support audio decoder plugins\n");
return false;
}
ffmpeg_audiodecoder = cvarfuncs->GetNVFDG("ffmpeg_audiodecoder_wip", "1", 0, "Enables the use of ffmpeg's decoder for pure audio files.", "ffmpeg");
if (!ffmpeg_audiodecoder->ival)
Con_Printf("ffmpeg: audio decoding disabled, use \"set %s 1\" to enable ffmpeg audio decoding\n", ffmpeg_audiodecoder->name);
return true;
}
//generic module stuff. this has to go somewhere.
static void AVLogCallback(void *avcl, int level, const char *fmt, va_list vl)
{ //needs to be reenterant
#ifdef _DEBUG
char string[1024];
Q_vsnprintf (string, sizeof(string), fmt, vl);
if (pdeveloper && pdeveloper->ival)
Con_Printf("%s", string);
#endif
}
//get the encoder/decoders to register themselves with the engine, then make sure avformat/avcodec have registered all they have to give.
qboolean AVEnc_Init(void);
qboolean AVDec_Init(void);
qboolean Plug_Init(void)
{
qboolean okay = false;
okay |= AVAudio_Init();
okay |= AVDec_Init();
okay |= AVEnc_Init();
if (okay)
{
#if ( LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(58,9,100) )
av_register_all();
avcodec_register_all();
#endif
pdeveloper = cvarfuncs->GetNVFDG("developer", "0", 0, "Developer spam.", "ffmpeg");
av_log_set_level(AV_LOG_WARNING);
av_log_set_callback(AVLogCallback);
}
return okay;
}