/* 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); }