/* RetroArch - A frontend for libretro.
* Copyright (C) 2010-2013 - Hans-Kristian Arntzen
*
* RetroArch is free software: you can redistribute it and/or modify it under the terms
* of the GNU General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along with RetroArch.
* If not, see .
*/
#include "../msvc/msvc_compat.h"
#ifdef __cplusplus
extern "C" {
#endif
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#ifdef __cplusplus
}
#endif
#include
#include
#include
#include "../boolean.h"
#include "../fifo_buffer.h"
#include "../thread.h"
#include "../general.h"
#include "../gfx/scaler/scaler.h"
#include "../conf/config_file.h"
#include "../audio/utils.h"
#include "../audio/resampler.h"
#include "ffemu.h"
#include
#ifdef FFEMU_PERF
#include
#endif
#ifdef HAVE_CONFIG_H
#include "../config.h"
#endif
struct ff_video_info
{
AVCodecContext *codec;
AVCodec *encoder;
AVFrame *conv_frame;
uint8_t *conv_frame_buf;
int64_t frame_cnt;
uint8_t *outbuf;
size_t outbuf_size;
// Output pixel format.
enum PixelFormat pix_fmt;
// Input pixel format. Only used by sws.
enum PixelFormat in_pix_fmt;
unsigned frame_drop_ratio;
unsigned frame_drop_count;
// Input pixel size.
size_t pix_size;
AVFormatContext *format;
struct scaler_ctx scaler;
struct SwsContext *sws;
bool use_sws;
};
struct ff_audio_info
{
AVCodecContext *codec;
AVCodec *encoder;
uint8_t *buffer;
size_t frames_in_buffer;
int64_t frame_cnt;
uint8_t *outbuf;
size_t outbuf_size;
// Most lossy audio codecs only support certain sampling rates.
// Could use libswresample, but it doesn't support floating point ratios. :(
// Use either S16 or (planar) float for simplicity.
const rarch_resampler_t *resampler;
void *resampler_data;
bool use_float;
bool is_planar;
unsigned sample_size;
float *float_conv;
size_t float_conv_frames;
float *resample_out;
size_t resample_out_frames;
int16_t *fixed_conv;
size_t fixed_conv_frames;
void *planar_buf;
size_t planar_buf_frames;
double ratio;
};
struct ff_muxer_info
{
AVFormatContext *ctx;
AVStream *astream;
AVStream *vstream;
};
struct ff_config_param
{
config_file_t *conf;
char vcodec[64];
char acodec[64];
char format[64];
enum PixelFormat out_pix_fmt;
unsigned threads;
unsigned frame_drop_ratio;
unsigned sample_rate;
unsigned scale_factor;
// Keep same naming conventions as libavcodec.
bool audio_qscale;
int audio_global_quality;
int audio_bit_rate;
AVDictionary *video_opts;
AVDictionary *audio_opts;
};
struct ffemu
{
struct ff_video_info video;
struct ff_audio_info audio;
struct ff_muxer_info muxer;
struct ff_config_param config;
struct ffemu_params params;
scond_t *cond;
slock_t *cond_lock;
slock_t *lock;
fifo_buffer_t *audio_fifo;
fifo_buffer_t *video_fifo;
fifo_buffer_t *attr_fifo;
sthread_t *thread;
volatile bool alive;
volatile bool can_sleep;
};
static bool ffemu_codec_has_sample_format(enum AVSampleFormat fmt, const enum AVSampleFormat *fmts)
{
unsigned i;
for (i = 0; fmts[i] != AV_SAMPLE_FMT_NONE; i++)
if (fmt == fmts[i])
return true;
return false;
}
static void ffemu_audio_resolve_format(struct ff_audio_info *audio, const AVCodec *codec)
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLTP, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_FLTP;
audio->use_float = true;
audio->is_planar = true;
RARCH_LOG("[FFmpeg]: Using sample format FLTP.\n");
}
else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLT, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
audio->use_float = true;
audio->is_planar = false;
RARCH_LOG("[FFmpeg]: Using sample format FLT.\n");
}
else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16P, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_S16P;
audio->use_float = false;
audio->is_planar = true;
RARCH_LOG("[FFmpeg]: Using sample format S16P.\n");
}
else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16, codec->sample_fmts))
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
audio->use_float = false;
audio->is_planar = false;
RARCH_LOG("[FFmpeg]: Using sample format S16.\n");
}
audio->sample_size = audio->use_float ? sizeof(float) : sizeof(int16_t);
}
static void ffemu_audio_resolve_sample_rate(ffemu_t *handle, const AVCodec *codec)
{
unsigned i;
struct ff_config_param *params = &handle->config;
struct ffemu_params *param = &handle->params;
// We'll have to force resampling to some supported sampling rate.
if (codec->supported_samplerates && !params->sample_rate)
{
int input_rate = (int)param->samplerate;
// Favor closest sampling rate, but always prefer ratio > 1.0.
int best_rate = codec->supported_samplerates[0];
int best_diff = best_rate - input_rate;
for (i = 1; codec->supported_samplerates[i]; i++)
{
int diff = codec->supported_samplerates[i] - input_rate;
bool better_rate;
if (best_diff < 0)
better_rate = diff > best_diff;
else
better_rate = diff >= 0 && diff < best_diff;
if (better_rate)
{
best_rate = codec->supported_samplerates[i];
best_diff = diff;
}
}
params->sample_rate = best_rate;
RARCH_LOG("[FFmpeg]: Using output sampling rate: %u.\n", best_rate);
}
}
static bool ffemu_init_audio(ffemu_t *handle)
{
struct ff_config_param *params = &handle->config;
struct ff_audio_info *audio = &handle->audio;
struct ffemu_params *param = &handle->params;
AVCodec *codec = avcodec_find_encoder_by_name(*params->acodec ? params->acodec : "flac");
if (!codec)
{
RARCH_ERR("[FFmpeg]: Cannot find acodec %s.\n", *params->acodec ? params->acodec : "flac");
return false;
}
audio->encoder = codec;
audio->codec = avcodec_alloc_context3(codec);
audio->codec->codec_type = AVMEDIA_TYPE_AUDIO;
audio->codec->channels = param->channels;
audio->codec->channel_layout = param->channels > 1 ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
ffemu_audio_resolve_format(audio, codec);
ffemu_audio_resolve_sample_rate(handle, codec);
if (params->sample_rate)
{
audio->ratio = (double)params->sample_rate / param->samplerate;
audio->codec->sample_rate = params->sample_rate;
audio->codec->time_base = av_d2q(1.0 / params->sample_rate, 1000000);
rarch_resampler_realloc(&audio->resampler_data,
&audio->resampler,
*g_settings.audio.resampler ? g_settings.audio.resampler : NULL,
audio->ratio);
}
else
{
audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
audio->codec->sample_rate = (int)roundf(param->samplerate);
audio->codec->time_base = av_d2q(1.0 / param->samplerate, 1000000);
}
if (params->audio_qscale)
{
audio->codec->flags |= CODEC_FLAG_QSCALE;
audio->codec->global_quality = params->audio_global_quality;
}
else if (params->audio_bit_rate)
audio->codec->bit_rate = params->audio_bit_rate;
// Allow experimental codecs.
audio->codec->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
if (handle->muxer.ctx->oformat->flags & AVFMT_GLOBALHEADER)
audio->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (avcodec_open2(audio->codec, codec, params->audio_opts ? ¶ms->audio_opts : NULL) != 0)
return false;
if (!audio->codec->frame_size) // If not set (PCM), just set something.
audio->codec->frame_size = 1024;
audio->buffer = (uint8_t*)av_malloc(
audio->codec->frame_size *
audio->codec->channels *
audio->sample_size);
//RARCH_LOG("[FFmpeg]: Audio frame size: %d.\n", audio->codec->frame_size);
if (!audio->buffer)
return false;
audio->outbuf_size = FF_MIN_BUFFER_SIZE;
audio->outbuf = (uint8_t*)av_malloc(audio->outbuf_size);
if (!audio->outbuf)
return false;
return true;
}
static bool ffemu_init_video(ffemu_t *handle)
{
struct ff_config_param *params = &handle->config;
struct ff_video_info *video = &handle->video;
struct ffemu_params *param = &handle->params;
AVCodec *codec = NULL;
if (*params->vcodec)
codec = avcodec_find_encoder_by_name(params->vcodec);
else
{
// By default, lossless video.
av_dict_set(¶ms->video_opts, "qp", "0", 0);
codec = avcodec_find_encoder_by_name("libx264rgb");
}
if (!codec)
{
RARCH_ERR("[FFmpeg]: Cannot find vcodec %s.\n", *params->vcodec ? params->vcodec : "libx264rgb");
return false;
}
video->encoder = codec;
// Don't use swscaler unless format is not something "in-house" scaler supports.
// libswscale doesn't scale RGB -> RGB correctly (goes via YUV first), and it's non-trivial to fix
// upstream as it's heavily geared towards YUV.
// If we're dealing with strange formats or YUV, just use libswscale.
if (params->out_pix_fmt != PIX_FMT_NONE)
{
video->pix_fmt = params->out_pix_fmt;
if (video->pix_fmt != PIX_FMT_BGR24 && video->pix_fmt != PIX_FMT_RGB32)
video->use_sws = true;
switch (video->pix_fmt)
{
case PIX_FMT_BGR24:
video->scaler.out_fmt = SCALER_FMT_BGR24;
break;
case PIX_FMT_RGB32:
video->scaler.out_fmt = SCALER_FMT_ARGB8888;
break;
default:
break;
}
}
else // Use BGR24 as default out format.
{
video->pix_fmt = PIX_FMT_BGR24;
video->scaler.out_fmt = SCALER_FMT_BGR24;
}
switch (param->pix_fmt)
{
case FFEMU_PIX_RGB565:
video->scaler.in_fmt = SCALER_FMT_RGB565;
video->in_pix_fmt = PIX_FMT_RGB565;
video->pix_size = 2;
break;
case FFEMU_PIX_BGR24:
video->scaler.in_fmt = SCALER_FMT_BGR24;
video->in_pix_fmt = PIX_FMT_BGR24;
video->pix_size = 3;
break;
case FFEMU_PIX_ARGB8888:
video->scaler.in_fmt = SCALER_FMT_ARGB8888;
video->in_pix_fmt = PIX_FMT_RGB32;
video->pix_size = 4;
break;
default:
return false;
}
video->codec = avcodec_alloc_context3(codec);
// Useful to set scale_factor to 2 for chroma subsampled formats to maintain full chroma resolution.
// (Or just use 4:4:4 or RGB ...)
param->out_width *= params->scale_factor;
param->out_height *= params->scale_factor;
video->codec->codec_type = AVMEDIA_TYPE_VIDEO;
video->codec->width = param->out_width;
video->codec->height = param->out_height;
video->codec->time_base = av_d2q((double)params->frame_drop_ratio / param->fps, 1000000); // Arbitrary big number.
video->codec->sample_aspect_ratio = av_d2q(param->aspect_ratio * param->out_height / param->out_width, 255);
video->codec->pix_fmt = video->pix_fmt;
video->codec->thread_count = params->threads;
if (handle->muxer.ctx->oformat->flags & AVFMT_GLOBALHEADER)
video->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (avcodec_open2(video->codec, codec, params->video_opts ? ¶ms->video_opts : NULL) != 0)
return false;
// Allocate a big buffer :p ffmpeg API doesn't seem to give us some clues how big this buffer should be.
video->outbuf_size = 1 << 23;
video->outbuf = (uint8_t*)av_malloc(video->outbuf_size);
video->frame_drop_ratio = params->frame_drop_ratio;
size_t size = avpicture_get_size(video->pix_fmt, param->out_width, param->out_height);
video->conv_frame_buf = (uint8_t*)av_malloc(size);
video->conv_frame = avcodec_alloc_frame();
avpicture_fill((AVPicture*)video->conv_frame, video->conv_frame_buf, video->pix_fmt,
param->out_width, param->out_height);
return true;
}
static bool ffemu_init_config(struct ff_config_param *params, const char *config)
{
params->out_pix_fmt = PIX_FMT_NONE;
params->scale_factor = 1;
params->threads = 1;
params->frame_drop_ratio = 1;
if (!config)
return true;
params->conf = config_file_new(config);
if (!params->conf)
{
RARCH_ERR("Failed to load FFmpeg config \"%s\".\n", config);
return false;
}
config_get_array(params->conf, "vcodec", params->vcodec, sizeof(params->vcodec));
config_get_array(params->conf, "acodec", params->acodec, sizeof(params->acodec));
config_get_array(params->conf, "format", params->format, sizeof(params->format));
config_get_uint(params->conf, "threads", ¶ms->threads);
if (!config_get_uint(params->conf, "frame_drop_ratio", ¶ms->frame_drop_ratio)
|| !params->frame_drop_ratio)
params->frame_drop_ratio = 1;
config_get_uint(params->conf, "sample_rate", ¶ms->sample_rate);
config_get_uint(params->conf, "scale_factor", ¶ms->scale_factor);
params->audio_qscale = config_get_int(params->conf, "audio_global_quality", ¶ms->audio_global_quality);
config_get_int(params->conf, "audio_bit_rate", ¶ms->audio_bit_rate);
char pix_fmt[64] = {0};
if (config_get_array(params->conf, "pix_fmt", pix_fmt, sizeof(pix_fmt)))
{
params->out_pix_fmt = av_get_pix_fmt(pix_fmt);
if (params->out_pix_fmt == PIX_FMT_NONE)
{
RARCH_ERR("Cannot find pix_fmt \"%s\".\n", pix_fmt);
return false;
}
}
struct config_file_entry entry;
if (!config_get_entry_list_head(params->conf, &entry))
return true;
do
{
if (strstr(entry.key, "video_") == entry.key)
{
const char *key = entry.key + strlen("video_");
av_dict_set(¶ms->video_opts, key, entry.value, 0);
}
else if (strstr(entry.key, "audio_") == entry.key)
{
const char *key = entry.key + strlen("audio_");
av_dict_set(¶ms->audio_opts, key, entry.value, 0);
}
} while (config_get_entry_list_next(&entry));
return true;
}
static bool ffemu_init_muxer_pre(ffemu_t *handle)
{
AVFormatContext *ctx = avformat_alloc_context();
av_strlcpy(ctx->filename, handle->params.filename, sizeof(ctx->filename));
if (*handle->config.format)
ctx->oformat = av_guess_format(handle->config.format, NULL, NULL);
else
ctx->oformat = av_guess_format(NULL, ctx->filename, NULL);
if (!ctx->oformat)
return false;
if (avio_open(&ctx->pb, ctx->filename, AVIO_FLAG_WRITE) < 0)
{
av_free(ctx);
return false;
}
handle->muxer.ctx = ctx;
return true;
}
static bool ffemu_init_muxer_post(ffemu_t *handle)
{
AVStream *stream = avformat_new_stream(handle->muxer.ctx, handle->video.encoder);
stream->codec = handle->video.codec;
handle->muxer.vstream = stream;
handle->muxer.vstream->sample_aspect_ratio = handle->video.codec->sample_aspect_ratio;
stream = avformat_new_stream(handle->muxer.ctx, handle->audio.encoder);
stream->codec = handle->audio.codec;
handle->muxer.astream = stream;
av_dict_set(&handle->muxer.ctx->metadata, "title", "RetroArch video dump", 0);
return avformat_write_header(handle->muxer.ctx, NULL) >= 0;
}
#define MAX_FRAMES 32
static void ffemu_thread(void *data);
static bool init_thread(ffemu_t *handle)
{
handle->lock = slock_new();
handle->cond_lock = slock_new();
handle->cond = scond_new();
handle->audio_fifo = fifo_new(32000 * sizeof(int16_t) * handle->params.channels * MAX_FRAMES / 60); // Some arbitrary max size.
handle->attr_fifo = fifo_new(sizeof(struct ffemu_video_data) * MAX_FRAMES);
handle->video_fifo = fifo_new(handle->params.fb_width * handle->params.fb_height *
handle->video.pix_size * MAX_FRAMES);
handle->alive = true;
handle->can_sleep = true;
handle->thread = sthread_create(ffemu_thread, handle);
assert(handle->lock && handle->cond_lock &&
handle->cond && handle->audio_fifo &&
handle->attr_fifo && handle->video_fifo && handle->thread);
return true;
}
static void deinit_thread(ffemu_t *handle)
{
if (!handle->thread)
return;
slock_lock(handle->cond_lock);
handle->alive = false;
handle->can_sleep = false;
slock_unlock(handle->cond_lock);
scond_signal(handle->cond);
sthread_join(handle->thread);
slock_free(handle->lock);
slock_free(handle->cond_lock);
scond_free(handle->cond);
handle->thread = NULL;
}
static void deinit_thread_buf(ffemu_t *handle)
{
if (handle->audio_fifo)
{
fifo_free(handle->audio_fifo);
handle->audio_fifo = NULL;
}
if (handle->attr_fifo)
{
fifo_free(handle->attr_fifo);
handle->attr_fifo = NULL;
}
if (handle->video_fifo)
{
fifo_free(handle->video_fifo);
handle->video_fifo = NULL;
}
}
ffemu_t *ffemu_new(const struct ffemu_params *params)
{
av_register_all();
avformat_network_init();
ffemu_t *handle = (ffemu_t*)calloc(1, sizeof(*handle));
if (!handle)
goto error;
handle->params = *params;
if (!ffemu_init_config(&handle->config, params->config))
goto error;
if (!ffemu_init_muxer_pre(handle))
goto error;
if (!ffemu_init_video(handle))
goto error;
if (!ffemu_init_audio(handle))
goto error;
if (!ffemu_init_muxer_post(handle))
goto error;
if (!init_thread(handle))
goto error;
return handle;
error:
ffemu_free(handle);
return NULL;
}
void ffemu_free(ffemu_t *handle)
{
if (!handle)
return;
deinit_thread(handle);
deinit_thread_buf(handle);
if (handle->audio.codec)
{
avcodec_close(handle->audio.codec);
av_free(handle->audio.codec);
}
av_free(handle->audio.buffer);
if (handle->video.codec)
{
avcodec_close(handle->video.codec);
av_free(handle->video.codec);
}
av_free(handle->video.conv_frame);
av_free(handle->video.conv_frame_buf);
scaler_ctx_gen_reset(&handle->video.scaler);
if (handle->video.sws)
sws_freeContext(handle->video.sws);
if (handle->config.conf)
config_file_free(handle->config.conf);
if (handle->config.video_opts)
av_dict_free(&handle->config.video_opts);
if (handle->config.audio_opts)
av_dict_free(&handle->config.audio_opts);
rarch_resampler_freep(&handle->audio.resampler,
&handle->audio.resampler_data);
av_free(handle->audio.float_conv);
av_free(handle->audio.resample_out);
av_free(handle->audio.fixed_conv);
av_free(handle->audio.planar_buf);
free(handle);
}
bool ffemu_push_video(ffemu_t *handle, const struct ffemu_video_data *data)
{
unsigned y;
bool drop_frame = handle->video.frame_drop_count++ % handle->video.frame_drop_ratio;
handle->video.frame_drop_count %= handle->video.frame_drop_ratio;
if (drop_frame)
return true;
for (;;)
{
slock_lock(handle->lock);
unsigned avail = fifo_write_avail(handle->attr_fifo);
slock_unlock(handle->lock);
if (!handle->alive)
return false;
if (avail >= sizeof(*data))
break;
slock_lock(handle->cond_lock);
if (handle->can_sleep)
{
handle->can_sleep = false;
scond_wait(handle->cond, handle->cond_lock);
handle->can_sleep = true;
}
else
scond_signal(handle->cond);
slock_unlock(handle->cond_lock);
}
slock_lock(handle->lock);
// Tightly pack our frame to conserve memory. libretro tends to use a very large pitch.
struct ffemu_video_data attr_data = *data;
if (attr_data.is_dupe)
attr_data.width = attr_data.height = attr_data.pitch = 0;
else
attr_data.pitch = attr_data.width * handle->video.pix_size;
fifo_write(handle->attr_fifo, &attr_data, sizeof(attr_data));
int offset = 0;
for (y = 0; y < attr_data.height; y++, offset += data->pitch)
fifo_write(handle->video_fifo, (const uint8_t*)data->data + offset, attr_data.pitch);
slock_unlock(handle->lock);
scond_signal(handle->cond);
return true;
}
bool ffemu_push_audio(ffemu_t *handle, const struct ffemu_audio_data *data)
{
for (;;)
{
slock_lock(handle->lock);
unsigned avail = fifo_write_avail(handle->audio_fifo);
slock_unlock(handle->lock);
if (!handle->alive)
return false;
if (avail >= data->frames * handle->params.channels * sizeof(int16_t))
break;
slock_lock(handle->cond_lock);
if (handle->can_sleep)
{
handle->can_sleep = false;
scond_wait(handle->cond, handle->cond_lock);
handle->can_sleep = true;
}
else
scond_signal(handle->cond);
slock_unlock(handle->cond_lock);
}
slock_lock(handle->lock);
fifo_write(handle->audio_fifo, data->data, data->frames * handle->params.channels * sizeof(int16_t));
slock_unlock(handle->lock);
scond_signal(handle->cond);
return true;
}
static bool encode_video(ffemu_t *handle, AVPacket *pkt, AVFrame *frame)
{
av_init_packet(pkt);
pkt->data = handle->video.outbuf;
pkt->size = handle->video.outbuf_size;
int got_packet = 0;
if (avcodec_encode_video2(handle->video.codec, pkt, frame, &got_packet) < 0)
return false;
if (!got_packet)
{
pkt->size = 0;
pkt->pts = AV_NOPTS_VALUE;
pkt->dts = AV_NOPTS_VALUE;
return true;
}
if (pkt->pts != (int64_t)AV_NOPTS_VALUE)
{
pkt->pts = av_rescale_q(pkt->pts, handle->video.codec->time_base,
handle->muxer.vstream->time_base);
}
if (pkt->dts != (int64_t)AV_NOPTS_VALUE)
{
pkt->dts = av_rescale_q(pkt->dts, handle->video.codec->time_base,
handle->muxer.vstream->time_base);
}
pkt->stream_index = handle->muxer.vstream->index;
return true;
}
static void ffemu_scale_input(ffemu_t *handle, const struct ffemu_video_data *data)
{
// Attempt to preserve more information if we scale down.
bool shrunk = handle->params.out_width < data->width || handle->params.out_height < data->height;
if (handle->video.use_sws)
{
handle->video.sws = sws_getCachedContext(handle->video.sws, data->width, data->height, handle->video.in_pix_fmt,
handle->params.out_width, handle->params.out_height, handle->video.pix_fmt,
shrunk ? SWS_BILINEAR : SWS_POINT, NULL, NULL, NULL);
int linesize = data->pitch;
sws_scale(handle->video.sws, (const uint8_t* const*)&data->data, &linesize, 0,
data->height, handle->video.conv_frame->data, handle->video.conv_frame->linesize);
}
else
{
if ((int)data->width != handle->video.scaler.in_width || (int)data->height != handle->video.scaler.in_height)
{
handle->video.scaler.in_width = data->width;
handle->video.scaler.in_height = data->height;
handle->video.scaler.in_stride = data->pitch;
handle->video.scaler.scaler_type = shrunk ? SCALER_TYPE_BILINEAR : SCALER_TYPE_POINT;
handle->video.scaler.out_width = handle->params.out_width;
handle->video.scaler.out_height = handle->params.out_height;
handle->video.scaler.out_stride = handle->video.conv_frame->linesize[0];
scaler_ctx_gen_filter(&handle->video.scaler);
}
scaler_ctx_scale(&handle->video.scaler, handle->video.conv_frame->data[0], data->data);
}
}
static bool ffemu_push_video_thread(ffemu_t *handle, const struct ffemu_video_data *data)
{
if (!data->is_dupe)
ffemu_scale_input(handle, data);
handle->video.conv_frame->pts = handle->video.frame_cnt;
AVPacket pkt;
if (!encode_video(handle, &pkt, handle->video.conv_frame))
return false;
if (pkt.size)
{
if (av_interleaved_write_frame(handle->muxer.ctx, &pkt) < 0)
return false;
}
handle->video.frame_cnt++;
return true;
}
static void planarize_float(float *out, const float *in, size_t frames)
{
size_t i;
for (i = 0; i < frames; i++)
{
out[i] = in[2 * i + 0];
out[i + frames] = in[2 * i + 1];
}
}
static void planarize_s16(int16_t *out, const int16_t *in, size_t frames)
{
size_t i;
for (i = 0; i < frames; i++)
{
out[i] = in[2 * i + 0];
out[i + frames] = in[2 * i + 1];
}
}
static void planarize_audio(ffemu_t *handle)
{
if (!handle->audio.is_planar)
return;
if (handle->audio.frames_in_buffer > handle->audio.planar_buf_frames)
{
handle->audio.planar_buf = av_realloc(handle->audio.planar_buf,
handle->audio.frames_in_buffer * handle->params.channels * handle->audio.sample_size);
if (!handle->audio.planar_buf)
return;
handle->audio.planar_buf_frames = handle->audio.frames_in_buffer;
}
if (handle->audio.use_float)
planarize_float((float*)handle->audio.planar_buf,
(const float*)handle->audio.buffer, handle->audio.frames_in_buffer);
else
planarize_s16((int16_t*)handle->audio.planar_buf,
(const int16_t*)handle->audio.buffer, handle->audio.frames_in_buffer);
}
static bool encode_audio(ffemu_t *handle, AVPacket *pkt, bool dry)
{
av_init_packet(pkt);
pkt->data = handle->audio.outbuf;
pkt->size = handle->audio.outbuf_size;
AVFrame *frame = avcodec_alloc_frame();
if (!frame)
return false;
frame->nb_samples = handle->audio.frames_in_buffer;
frame->format = handle->audio.codec->sample_fmt;
frame->channel_layout = handle->audio.codec->channel_layout;
frame->pts = handle->audio.frame_cnt;
planarize_audio(handle);
int samples_size = av_samples_get_buffer_size(NULL, handle->audio.codec->channels,
handle->audio.frames_in_buffer,
handle->audio.codec->sample_fmt, 0);
avcodec_fill_audio_frame(frame, handle->audio.codec->channels,
handle->audio.codec->sample_fmt,
handle->audio.is_planar ? (uint8_t*)handle->audio.planar_buf : handle->audio.buffer,
samples_size, 0);
int got_packet = 0;
if (avcodec_encode_audio2(handle->audio.codec,
pkt, dry ? NULL : frame, &got_packet) < 0)
{
avcodec_free_frame(&frame);
return false;
}
if (!got_packet)
{
pkt->size = 0;
pkt->pts = AV_NOPTS_VALUE;
pkt->dts = AV_NOPTS_VALUE;
avcodec_free_frame(&frame);
return true;
}
if (pkt->pts != (int64_t)AV_NOPTS_VALUE)
{
pkt->pts = av_rescale_q(pkt->pts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
if (pkt->dts != (int64_t)AV_NOPTS_VALUE)
{
pkt->dts = av_rescale_q(pkt->dts,
handle->audio.codec->time_base,
handle->muxer.astream->time_base);
}
avcodec_free_frame(&frame);
pkt->stream_index = handle->muxer.astream->index;
return true;
}
static void ffemu_audio_resample(ffemu_t *handle, struct ffemu_audio_data *data)
{
if (!handle->audio.use_float && !handle->audio.resampler)
return;
if (data->frames > handle->audio.float_conv_frames)
{
handle->audio.float_conv = (float*)av_realloc(handle->audio.float_conv,
data->frames * handle->params.channels * sizeof(float));
if (!handle->audio.float_conv)
return;
handle->audio.float_conv_frames = data->frames;
// To make sure we don't accidentially overflow.
handle->audio.resample_out_frames = data->frames * handle->audio.ratio + 16;
handle->audio.resample_out = (float*)av_realloc(handle->audio.resample_out,
handle->audio.resample_out_frames * handle->params.channels * sizeof(float));
if (!handle->audio.resample_out)
return;
handle->audio.fixed_conv_frames = max(handle->audio.resample_out_frames, handle->audio.float_conv_frames);
handle->audio.fixed_conv = (int16_t*)av_realloc(handle->audio.fixed_conv,
handle->audio.fixed_conv_frames * handle->params.channels * sizeof(int16_t));
if (!handle->audio.fixed_conv)
return;
}
if (handle->audio.use_float || handle->audio.resampler)
{
audio_convert_s16_to_float(handle->audio.float_conv,
(const int16_t*)data->data, data->frames * handle->params.channels, 1.0);
data->data = handle->audio.float_conv;
}
if (handle->audio.resampler)
{
// It's always two channels ...
struct resampler_data info = {0};
info.data_in = (const float*)data->data;
info.data_out = handle->audio.resample_out;
info.input_frames = data->frames;
info.ratio = handle->audio.ratio;
rarch_resampler_process(handle->audio.resampler, handle->audio.resampler_data, &info);
data->data = handle->audio.resample_out;
data->frames = info.output_frames;
if (!handle->audio.use_float)
{
audio_convert_float_to_s16(handle->audio.fixed_conv, handle->audio.resample_out,
data->frames * handle->params.channels);
data->data = handle->audio.fixed_conv;
}
}
}
static bool ffemu_push_audio_thread(ffemu_t *handle, struct ffemu_audio_data *data, bool require_block)
{
ffemu_audio_resample(handle, data);
size_t written_frames = 0;
while (written_frames < data->frames)
{
size_t can_write = handle->audio.codec->frame_size - handle->audio.frames_in_buffer;
size_t write_left = data->frames - written_frames;
size_t write_frames = write_left > can_write ? can_write : write_left;
size_t write_size = write_frames * handle->params.channels * handle->audio.sample_size;
size_t bytes_in_buffer = handle->audio.frames_in_buffer * handle->params.channels * handle->audio.sample_size;
size_t written_bytes = written_frames * handle->params.channels * handle->audio.sample_size;
memcpy(handle->audio.buffer + bytes_in_buffer,
(const uint8_t*)data->data + written_bytes,
write_size);
written_frames += write_frames;
handle->audio.frames_in_buffer += write_frames;
if ((handle->audio.frames_in_buffer < (size_t)handle->audio.codec->frame_size) && require_block)
break;
AVPacket pkt;
if (!encode_audio(handle, &pkt, false))
return false;
handle->audio.frame_cnt += handle->audio.frames_in_buffer;
handle->audio.frames_in_buffer = 0;
if (pkt.size)
{
if (av_interleaved_write_frame(handle->muxer.ctx, &pkt) < 0)
return false;
}
}
return true;
}
static void ffemu_flush_audio(ffemu_t *handle, void *audio_buf, size_t audio_buf_size)
{
size_t avail = fifo_read_avail(handle->audio_fifo);
if (avail)
{
fifo_read(handle->audio_fifo, audio_buf, avail);
struct ffemu_audio_data aud = {0};
aud.frames = avail / (sizeof(int16_t) * handle->params.channels);
aud.data = audio_buf;
ffemu_push_audio_thread(handle, &aud, false);
}
for (;;)
{
AVPacket pkt;
if (!encode_audio(handle, &pkt, true) || !pkt.size ||
av_interleaved_write_frame(handle->muxer.ctx, &pkt) < 0)
break;
}
}
static void ffemu_flush_video(ffemu_t *handle)
{
for (;;)
{
AVPacket pkt;
if (!encode_video(handle, &pkt, NULL) || !pkt.size ||
av_interleaved_write_frame(handle->muxer.ctx, &pkt) < 0)
break;
}
}
static void ffemu_flush_buffers(ffemu_t *handle)
{
void *video_buf = av_malloc(2 * handle->params.fb_width * handle->params.fb_height * handle->video.pix_size);
size_t audio_buf_size = handle->audio.codec->frame_size * handle->params.channels * sizeof(int16_t);
void *audio_buf = av_malloc(audio_buf_size);
// Try pushing data in an interleaving pattern to ease the work of the muxer a bit.
bool did_work;
do
{
did_work = false;
if (fifo_read_avail(handle->audio_fifo) >= audio_buf_size)
{
fifo_read(handle->audio_fifo, audio_buf, audio_buf_size);
struct ffemu_audio_data aud = {0};
aud.frames = handle->audio.codec->frame_size;
aud.data = audio_buf;
ffemu_push_audio_thread(handle, &aud, true);
did_work = true;
}
struct ffemu_video_data attr_buf;
if (fifo_read_avail(handle->attr_fifo) >= sizeof(attr_buf))
{
fifo_read(handle->attr_fifo, &attr_buf, sizeof(attr_buf));
fifo_read(handle->video_fifo, video_buf, attr_buf.height * attr_buf.pitch);
attr_buf.data = video_buf;
ffemu_push_video_thread(handle, &attr_buf);
did_work = true;
}
} while (did_work);
// Flush out last audio.
ffemu_flush_audio(handle, audio_buf, audio_buf_size);
// Flush out last video.
ffemu_flush_video(handle);
av_free(video_buf);
av_free(audio_buf);
}
bool ffemu_finalize(ffemu_t *handle)
{
deinit_thread(handle);
// Flush out data still in buffers (internal, and FFmpeg internal).
ffemu_flush_buffers(handle);
deinit_thread_buf(handle);
// Write final data.
av_write_trailer(handle->muxer.ctx);
return true;
}
static void ffemu_thread(void *data)
{
ffemu_t *ff = (ffemu_t*)data;
// For some reason, FFmpeg has a tendency to crash if we don't overallocate a bit. :s
void *video_buf = av_malloc(2 * ff->params.fb_width * ff->params.fb_height * ff->video.pix_size);
assert(video_buf);
size_t audio_buf_size = ff->audio.codec->frame_size * ff->params.channels * sizeof(int16_t);
void *audio_buf = av_malloc(audio_buf_size);
while (ff->alive)
{
struct ffemu_video_data attr_buf;
bool avail_video = false;
bool avail_audio = false;
slock_lock(ff->lock);
if (fifo_read_avail(ff->attr_fifo) >= sizeof(attr_buf))
avail_video = true;
if (fifo_read_avail(ff->audio_fifo) >= audio_buf_size)
avail_audio = true;
slock_unlock(ff->lock);
if (!avail_video && !avail_audio)
{
slock_lock(ff->cond_lock);
if (ff->can_sleep)
{
ff->can_sleep = false;
scond_wait(ff->cond, ff->cond_lock);
ff->can_sleep = true;
}
else
scond_signal(ff->cond);
slock_unlock(ff->cond_lock);
}
if (avail_video)
{
slock_lock(ff->lock);
fifo_read(ff->attr_fifo, &attr_buf, sizeof(attr_buf));
fifo_read(ff->video_fifo, video_buf, attr_buf.height * attr_buf.pitch);
slock_unlock(ff->lock);
scond_signal(ff->cond);
attr_buf.data = video_buf;
ffemu_push_video_thread(ff, &attr_buf);
}
if (avail_audio)
{
slock_lock(ff->lock);
fifo_read(ff->audio_fifo, audio_buf, audio_buf_size);
slock_unlock(ff->lock);
scond_signal(ff->cond);
struct ffemu_audio_data aud = {0};
aud.frames = ff->audio.codec->frame_size;
aud.data = audio_buf;
ffemu_push_audio_thread(ff, &aud, true);
}
}
av_free(video_buf);
av_free(audio_buf);
}