mirror of
https://github.com/libretro/RetroArch.git
synced 2024-11-28 02:30:35 +00:00
1259 lines
35 KiB
C
1259 lines
35 KiB
C
/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
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*
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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* ation, either version 3 of the License, or (at your option) any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
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* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along with RetroArch.
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* If not, see <http://www.gnu.org/licenses/>.
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*/
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#include "../msvc/msvc_compat.h"
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#ifdef __cplusplus
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extern "C" {
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#endif
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#include <libavcodec/avcodec.h>
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#include <libavutil/mathematics.h>
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#include <libavutil/avutil.h>
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#include <libavutil/avstring.h>
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#include <libavutil/opt.h>
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#include <libavformat/avformat.h>
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#include <libavutil/avconfig.h>
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#include <libavutil/pixdesc.h>
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#include <libavutil/channel_layout.h>
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#include <libswscale/swscale.h>
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#ifdef __cplusplus
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}
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#endif
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#include <stdint.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include "../boolean.h"
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#include "../fifo_buffer.h"
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#include "../thread.h"
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#include "../general.h"
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#include "../gfx/scaler/scaler.h"
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#include "../conf/config_file.h"
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#include "../audio/utils.h"
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#include "../audio/resampler.h"
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#include "ffemu.h"
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#include <assert.h>
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#ifdef FFEMU_PERF
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#include <time.h>
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#endif
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#ifdef HAVE_CONFIG_H
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#include "../config.h"
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#endif
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struct ff_video_info
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{
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AVCodecContext *codec;
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AVCodec *encoder;
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AVFrame *conv_frame;
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uint8_t *conv_frame_buf;
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int64_t frame_cnt;
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uint8_t *outbuf;
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size_t outbuf_size;
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// Output pixel format.
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enum PixelFormat pix_fmt;
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// Input pixel format. Only used by sws.
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enum PixelFormat in_pix_fmt;
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unsigned frame_drop_ratio;
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unsigned frame_drop_count;
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// Input pixel size.
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size_t pix_size;
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AVFormatContext *format;
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struct scaler_ctx scaler;
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struct SwsContext *sws;
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bool use_sws;
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};
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struct ff_audio_info
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{
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AVCodecContext *codec;
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AVCodec *encoder;
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uint8_t *buffer;
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size_t frames_in_buffer;
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int64_t frame_cnt;
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uint8_t *outbuf;
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size_t outbuf_size;
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// Most lossy audio codecs only support certain sampling rates.
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// Could use libswresample, but it doesn't support floating point ratios. :(
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// Use either S16 or (planar) float for simplicity.
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const rarch_resampler_t *resampler;
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void *resampler_data;
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bool use_float;
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bool is_planar;
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unsigned sample_size;
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float *float_conv;
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size_t float_conv_frames;
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float *resample_out;
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size_t resample_out_frames;
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int16_t *fixed_conv;
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size_t fixed_conv_frames;
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void *planar_buf;
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size_t planar_buf_frames;
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double ratio;
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};
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struct ff_muxer_info
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{
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AVFormatContext *ctx;
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AVStream *astream;
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AVStream *vstream;
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};
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struct ff_config_param
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{
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config_file_t *conf;
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char vcodec[64];
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char acodec[64];
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char format[64];
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enum PixelFormat out_pix_fmt;
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unsigned threads;
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unsigned frame_drop_ratio;
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unsigned sample_rate;
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unsigned scale_factor;
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// Keep same naming conventions as libavcodec.
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bool audio_qscale;
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int audio_global_quality;
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int audio_bit_rate;
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AVDictionary *video_opts;
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AVDictionary *audio_opts;
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};
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struct ffemu
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{
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struct ff_video_info video;
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struct ff_audio_info audio;
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struct ff_muxer_info muxer;
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struct ff_config_param config;
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struct ffemu_params params;
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scond_t *cond;
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slock_t *cond_lock;
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slock_t *lock;
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fifo_buffer_t *audio_fifo;
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fifo_buffer_t *video_fifo;
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fifo_buffer_t *attr_fifo;
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sthread_t *thread;
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volatile bool alive;
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volatile bool can_sleep;
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};
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static bool ffemu_codec_has_sample_format(enum AVSampleFormat fmt, const enum AVSampleFormat *fmts)
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{
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unsigned i;
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for (i = 0; fmts[i] != AV_SAMPLE_FMT_NONE; i++)
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if (fmt == fmts[i])
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return true;
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return false;
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}
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static void ffemu_audio_resolve_format(struct ff_audio_info *audio, const AVCodec *codec)
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
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if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLTP, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_FLTP;
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audio->use_float = true;
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audio->is_planar = true;
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RARCH_LOG("[FFmpeg]: Using sample format FLTP.\n");
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}
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else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_FLT, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
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audio->use_float = true;
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audio->is_planar = false;
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RARCH_LOG("[FFmpeg]: Using sample format FLT.\n");
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}
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else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16P, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_S16P;
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audio->use_float = false;
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audio->is_planar = true;
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RARCH_LOG("[FFmpeg]: Using sample format S16P.\n");
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}
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else if (ffemu_codec_has_sample_format(AV_SAMPLE_FMT_S16, codec->sample_fmts))
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
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audio->use_float = false;
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audio->is_planar = false;
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RARCH_LOG("[FFmpeg]: Using sample format S16.\n");
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}
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audio->sample_size = audio->use_float ? sizeof(float) : sizeof(int16_t);
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}
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static void ffemu_audio_resolve_sample_rate(ffemu_t *handle, const AVCodec *codec)
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{
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unsigned i;
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struct ff_config_param *params = &handle->config;
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struct ffemu_params *param = &handle->params;
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// We'll have to force resampling to some supported sampling rate.
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if (codec->supported_samplerates && !params->sample_rate)
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{
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int input_rate = (int)param->samplerate;
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// Favor closest sampling rate, but always prefer ratio > 1.0.
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int best_rate = codec->supported_samplerates[0];
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int best_diff = best_rate - input_rate;
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for (i = 1; codec->supported_samplerates[i]; i++)
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{
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int diff = codec->supported_samplerates[i] - input_rate;
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bool better_rate;
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if (best_diff < 0)
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better_rate = diff > best_diff;
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else
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better_rate = diff >= 0 && diff < best_diff;
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if (better_rate)
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{
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best_rate = codec->supported_samplerates[i];
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best_diff = diff;
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}
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}
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params->sample_rate = best_rate;
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RARCH_LOG("[FFmpeg]: Using output sampling rate: %u.\n", best_rate);
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}
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}
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static bool ffemu_init_audio(ffemu_t *handle)
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{
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struct ff_config_param *params = &handle->config;
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struct ff_audio_info *audio = &handle->audio;
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struct ffemu_params *param = &handle->params;
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AVCodec *codec = avcodec_find_encoder_by_name(*params->acodec ? params->acodec : "flac");
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if (!codec)
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{
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RARCH_ERR("[FFmpeg]: Cannot find acodec %s.\n", *params->acodec ? params->acodec : "flac");
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return false;
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}
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audio->encoder = codec;
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audio->codec = avcodec_alloc_context3(codec);
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audio->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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audio->codec->channels = param->channels;
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audio->codec->channel_layout = param->channels > 1 ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
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ffemu_audio_resolve_format(audio, codec);
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ffemu_audio_resolve_sample_rate(handle, codec);
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if (params->sample_rate)
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{
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audio->ratio = (double)params->sample_rate / param->samplerate;
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audio->codec->sample_rate = params->sample_rate;
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audio->codec->time_base = av_d2q(1.0 / params->sample_rate, 1000000);
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rarch_resampler_realloc(&audio->resampler_data,
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&audio->resampler,
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*g_settings.audio.resampler ? g_settings.audio.resampler : NULL,
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audio->ratio);
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}
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else
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{
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audio->codec->sample_fmt = AV_SAMPLE_FMT_S16;
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audio->codec->sample_rate = (int)roundf(param->samplerate);
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audio->codec->time_base = av_d2q(1.0 / param->samplerate, 1000000);
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}
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if (params->audio_qscale)
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{
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audio->codec->flags |= CODEC_FLAG_QSCALE;
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audio->codec->global_quality = params->audio_global_quality;
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}
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else if (params->audio_bit_rate)
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audio->codec->bit_rate = params->audio_bit_rate;
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// Allow experimental codecs.
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audio->codec->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
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if (handle->muxer.ctx->oformat->flags & AVFMT_GLOBALHEADER)
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audio->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
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if (avcodec_open2(audio->codec, codec, params->audio_opts ? ¶ms->audio_opts : NULL) != 0)
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return false;
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if (!audio->codec->frame_size) // If not set (PCM), just set something.
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audio->codec->frame_size = 1024;
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audio->buffer = (uint8_t*)av_malloc(
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audio->codec->frame_size *
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audio->codec->channels *
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audio->sample_size);
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//RARCH_LOG("[FFmpeg]: Audio frame size: %d.\n", audio->codec->frame_size);
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if (!audio->buffer)
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return false;
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audio->outbuf_size = FF_MIN_BUFFER_SIZE;
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audio->outbuf = (uint8_t*)av_malloc(audio->outbuf_size);
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if (!audio->outbuf)
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return false;
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return true;
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}
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static bool ffemu_init_video(ffemu_t *handle)
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{
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struct ff_config_param *params = &handle->config;
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struct ff_video_info *video = &handle->video;
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struct ffemu_params *param = &handle->params;
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AVCodec *codec = NULL;
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if (*params->vcodec)
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codec = avcodec_find_encoder_by_name(params->vcodec);
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else
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{
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// By default, lossless video.
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av_dict_set(¶ms->video_opts, "qp", "0", 0);
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codec = avcodec_find_encoder_by_name("libx264rgb");
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}
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if (!codec)
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{
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RARCH_ERR("[FFmpeg]: Cannot find vcodec %s.\n", *params->vcodec ? params->vcodec : "libx264rgb");
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return false;
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}
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video->encoder = codec;
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// Don't use swscaler unless format is not something "in-house" scaler supports.
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// libswscale doesn't scale RGB -> RGB correctly (goes via YUV first), and it's non-trivial to fix
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// upstream as it's heavily geared towards YUV.
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// If we're dealing with strange formats or YUV, just use libswscale.
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if (params->out_pix_fmt != PIX_FMT_NONE)
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{
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video->pix_fmt = params->out_pix_fmt;
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if (video->pix_fmt != PIX_FMT_BGR24 && video->pix_fmt != PIX_FMT_RGB32)
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video->use_sws = true;
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switch (video->pix_fmt)
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{
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case PIX_FMT_BGR24:
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video->scaler.out_fmt = SCALER_FMT_BGR24;
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break;
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case PIX_FMT_RGB32:
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video->scaler.out_fmt = SCALER_FMT_ARGB8888;
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break;
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default:
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break;
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}
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}
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else // Use BGR24 as default out format.
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{
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video->pix_fmt = PIX_FMT_BGR24;
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video->scaler.out_fmt = SCALER_FMT_BGR24;
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}
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switch (param->pix_fmt)
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{
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case FFEMU_PIX_RGB565:
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video->scaler.in_fmt = SCALER_FMT_RGB565;
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video->in_pix_fmt = PIX_FMT_RGB565;
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video->pix_size = 2;
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break;
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case FFEMU_PIX_BGR24:
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video->scaler.in_fmt = SCALER_FMT_BGR24;
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video->in_pix_fmt = PIX_FMT_BGR24;
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video->pix_size = 3;
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break;
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case FFEMU_PIX_ARGB8888:
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video->scaler.in_fmt = SCALER_FMT_ARGB8888;
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video->in_pix_fmt = PIX_FMT_RGB32;
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video->pix_size = 4;
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break;
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default:
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return false;
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}
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video->codec = avcodec_alloc_context3(codec);
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// Useful to set scale_factor to 2 for chroma subsampled formats to maintain full chroma resolution.
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// (Or just use 4:4:4 or RGB ...)
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param->out_width *= params->scale_factor;
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param->out_height *= params->scale_factor;
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video->codec->codec_type = AVMEDIA_TYPE_VIDEO;
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video->codec->width = param->out_width;
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video->codec->height = param->out_height;
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video->codec->time_base = av_d2q((double)params->frame_drop_ratio / param->fps, 1000000); // Arbitrary big number.
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video->codec->sample_aspect_ratio = av_d2q(param->aspect_ratio * param->out_height / param->out_width, 255);
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video->codec->pix_fmt = video->pix_fmt;
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video->codec->thread_count = params->threads;
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if (handle->muxer.ctx->oformat->flags & AVFMT_GLOBALHEADER)
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video->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
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if (avcodec_open2(video->codec, codec, params->video_opts ? ¶ms->video_opts : NULL) != 0)
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return false;
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// Allocate a big buffer :p ffmpeg API doesn't seem to give us some clues how big this buffer should be.
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video->outbuf_size = 1 << 23;
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video->outbuf = (uint8_t*)av_malloc(video->outbuf_size);
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video->frame_drop_ratio = params->frame_drop_ratio;
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size_t size = avpicture_get_size(video->pix_fmt, param->out_width, param->out_height);
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video->conv_frame_buf = (uint8_t*)av_malloc(size);
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video->conv_frame = avcodec_alloc_frame();
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avpicture_fill((AVPicture*)video->conv_frame, video->conv_frame_buf, video->pix_fmt,
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param->out_width, param->out_height);
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return true;
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}
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static bool ffemu_init_config(struct ff_config_param *params, const char *config)
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{
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params->out_pix_fmt = PIX_FMT_NONE;
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params->scale_factor = 1;
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params->threads = 1;
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params->frame_drop_ratio = 1;
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if (!config)
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return true;
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params->conf = config_file_new(config);
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if (!params->conf)
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{
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RARCH_ERR("Failed to load FFmpeg config \"%s\".\n", config);
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return false;
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}
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config_get_array(params->conf, "vcodec", params->vcodec, sizeof(params->vcodec));
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config_get_array(params->conf, "acodec", params->acodec, sizeof(params->acodec));
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config_get_array(params->conf, "format", params->format, sizeof(params->format));
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config_get_uint(params->conf, "threads", ¶ms->threads);
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if (!config_get_uint(params->conf, "frame_drop_ratio", ¶ms->frame_drop_ratio)
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|| !params->frame_drop_ratio)
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params->frame_drop_ratio = 1;
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config_get_uint(params->conf, "sample_rate", ¶ms->sample_rate);
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config_get_uint(params->conf, "scale_factor", ¶ms->scale_factor);
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params->audio_qscale = config_get_int(params->conf, "audio_global_quality", ¶ms->audio_global_quality);
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config_get_int(params->conf, "audio_bit_rate", ¶ms->audio_bit_rate);
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char pix_fmt[64] = {0};
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if (config_get_array(params->conf, "pix_fmt", pix_fmt, sizeof(pix_fmt)))
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{
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params->out_pix_fmt = av_get_pix_fmt(pix_fmt);
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if (params->out_pix_fmt == PIX_FMT_NONE)
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{
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RARCH_ERR("Cannot find pix_fmt \"%s\".\n", pix_fmt);
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return false;
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}
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}
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struct config_file_entry entry;
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if (!config_get_entry_list_head(params->conf, &entry))
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return true;
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do
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{
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if (strstr(entry.key, "video_") == entry.key)
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|
{
|
|
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);
|
|
}
|
|
|