From d4bdd0473d6f43132257c9fb3848d829755167a3 Mon Sep 17 00:00:00 2001 From: wm4 Date: Mon, 5 Nov 2012 17:02:04 +0100 Subject: Rename directories, move files (step 1 of 2) (does not compile) Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code. --- audio/decode/ad_spdif.c | 310 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 310 insertions(+) create mode 100644 audio/decode/ad_spdif.c (limited to 'audio/decode/ad_spdif.c') diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c new file mode 100644 index 0000000000..877bc99317 --- /dev/null +++ b/audio/decode/ad_spdif.c @@ -0,0 +1,310 @@ +/* + * This file is part of MPlayer. + * + * MPlayer 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 Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include + +#include +#include +#include + +#include "config.h" +#include "mp_msg.h" +#include "ad_internal.h" + +static const ad_info_t info = { + "libavformat/spdifenc audio pass-through decoder.", + "spdif", + "Naoya OYAMA", + "Naoya OYAMA", + "For ALL hardware decoders" +}; + +LIBAD_EXTERN(spdif) + +#define FILENAME_SPDIFENC "spdif" +#define OUTBUF_SIZE 65536 +struct spdifContext { + AVFormatContext *lavf_ctx; + int iec61937_packet_size; + int out_buffer_len; + int out_buffer_size; + uint8_t *out_buffer; + uint8_t pb_buffer[OUTBUF_SIZE]; +}; + +static int read_packet(void *p, uint8_t *buf, int buf_size) +{ + // spdifenc does not use read callback. + return 0; +} + +static int write_packet(void *p, uint8_t *buf, int buf_size) +{ + int len; + struct spdifContext *ctx = p; + + len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len); + memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len); + ctx->out_buffer_len += len; + return len; +} + +static int64_t seek(void *p, int64_t offset, int whence) +{ + // spdifenc does not use seek callback. + return 0; +} + +static int preinit(sh_audio_t *sh) +{ + sh->samplesize = 2; + return 1; +} + +static int init(sh_audio_t *sh) +{ + int i, x, in_size, srate, bps, *dtshd_rate; + unsigned char *start; + double pts; + static const struct { + const char *name; enum CodecID id; + } fmt_id_type[] = { + { "aac" , CODEC_ID_AAC }, + { "ac3" , CODEC_ID_AC3 }, + { "dca" , CODEC_ID_DTS }, + { "eac3", CODEC_ID_EAC3 }, + { "mpa" , CODEC_ID_MP3 }, + { "thd" , CODEC_ID_TRUEHD }, + { NULL , 0 } + }; + AVFormatContext *lavf_ctx = NULL; + AVStream *stream = NULL; + const AVOption *opt = NULL; + struct spdifContext *spdif_ctx = NULL; + + spdif_ctx = av_mallocz(sizeof(*spdif_ctx)); + if (!spdif_ctx) + goto fail; + spdif_ctx->lavf_ctx = avformat_alloc_context(); + if (!spdif_ctx->lavf_ctx) + goto fail; + + sh->context = spdif_ctx; + lavf_ctx = spdif_ctx->lavf_ctx; + + lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL); + if (!lavf_ctx->oformat) + goto fail; + lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size); + if (!lavf_ctx->priv_data) + goto fail; + lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx, + read_packet, write_packet, seek); + if (!lavf_ctx->pb) + goto fail; + stream = avformat_new_stream(lavf_ctx, 0); + if (!stream) + goto fail; + lavf_ctx->duration = AV_NOPTS_VALUE; + lavf_ctx->start_time = AV_NOPTS_VALUE; + for (i = 0; fmt_id_type[i].name; i++) { + if (!strcmp(sh->codec->dll, fmt_id_type[i].name)) { + lavf_ctx->streams[0]->codec->codec_id = fmt_id_type[i].id; + break; + } + } + lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE; + if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) { + mp_msg(MSGT_DECAUDIO,MSGL_INFO, + "This codec is not supported by spdifenc.\n"); + goto fail; + } + + // get sample_rate & bitrate from parser + bps = srate = 0; + x = ds_get_packet_pts(sh->ds, &start, &pts); + in_size = x; + if (x <= 0) { + pts = MP_NOPTS_VALUE; + x = 0; + } + ds_parse(sh->ds, &start, &x, pts, 0); + if (x == 0) { // not enough buffer + srate = 48000; //fake value + bps = 768000/8; //fake value + } else if (sh->avctx) { + if (sh->avctx->sample_rate < 44100) { + mp_msg(MSGT_DECAUDIO,MSGL_INFO, + "This stream sample_rate[%d Hz] may be broken. " + "Force reset 48000Hz.\n", + sh->avctx->sample_rate); + srate = 48000; //fake value + } else + srate = sh->avctx->sample_rate; + bps = sh->avctx->bit_rate/8; + } + sh->ds->buffer_pos -= in_size; + + switch (lavf_ctx->streams[0]->codec->codec_id) { + case CODEC_ID_AAC: + spdif_ctx->iec61937_packet_size = 16384; + sh->sample_format = AF_FORMAT_IEC61937_LE; + sh->samplerate = srate; + sh->channels = 2; + sh->i_bps = bps; + break; + case CODEC_ID_AC3: + spdif_ctx->iec61937_packet_size = 6144; + sh->sample_format = AF_FORMAT_IEC61937_LE; + sh->samplerate = srate; + sh->channels = 2; + sh->i_bps = bps; + break; + case CODEC_ID_DTS: // FORCE USE DTS-HD + opt = av_opt_find(&lavf_ctx->oformat->priv_class, + "dtshd_rate", NULL, 0, 0); + if (!opt) + goto fail; + dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) + + opt->offset); + *dtshd_rate = 192000*4; + spdif_ctx->iec61937_packet_size = 32768; + sh->sample_format = AF_FORMAT_IEC61937_LE; + sh->samplerate = 192000; // DTS core require 48000 + sh->channels = 2*4; + sh->i_bps = bps; + break; + case CODEC_ID_EAC3: + spdif_ctx->iec61937_packet_size = 24576; + sh->sample_format = AF_FORMAT_IEC61937_LE; + sh->samplerate = 192000; + sh->channels = 2; + sh->i_bps = bps; + break; + case CODEC_ID_MP3: + spdif_ctx->iec61937_packet_size = 4608; + sh->sample_format = AF_FORMAT_MPEG2; + sh->samplerate = srate; + sh->channels = 2; + sh->i_bps = bps; + break; + case CODEC_ID_TRUEHD: + spdif_ctx->iec61937_packet_size = 61440; + sh->sample_format = AF_FORMAT_IEC61937_LE; + sh->samplerate = 192000; + sh->channels = 8; + sh->i_bps = bps; + break; + default: + break; + } + + return 1; + +fail: + uninit(sh); + return 0; +} + +static int decode_audio(sh_audio_t *sh, unsigned char *buf, + int minlen, int maxlen) +{ + struct spdifContext *spdif_ctx = sh->context; + AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx; + AVPacket pkt; + double pts; + int ret, in_size, consumed, x; + unsigned char *start = NULL; + + consumed = spdif_ctx->out_buffer_len = 0; + spdif_ctx->out_buffer_size = maxlen; + spdif_ctx->out_buffer = buf; + while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen + && spdif_ctx->out_buffer_len < minlen) { + if (sh->ds->eof) + break; + x = ds_get_packet_pts(sh->ds, &start, &pts); + if (x <= 0) { + x = 0; + ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0); + if (x == 0) + continue; // END_NOT_FOUND + in_size = x; + } else { + in_size = x; + consumed = ds_parse(sh->ds, &start, &x, pts, 0); + if (x == 0) { + mp_msg(MSGT_DECAUDIO,MSGL_V, + "start[%p] in_size[%d] consumed[%d] x[%d].\n", + start, in_size, consumed, x); + continue; // END_NOT_FOUND + } + sh->ds->buffer_pos -= in_size - consumed; + } + av_init_packet(&pkt); + pkt.data = start; + pkt.size = x; + mp_msg(MSGT_DECAUDIO,MSGL_V, + "start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n", + start, pkt.size, in_size, consumed, x); + if (pts != MP_NOPTS_VALUE) { + sh->pts = pts; + sh->pts_bytes = 0; + } + ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt); + if (ret < 0) + break; + } + sh->pts_bytes += spdif_ctx->out_buffer_len; + return spdif_ctx->out_buffer_len; +} + +static int control(sh_audio_t *sh, int cmd, void* arg, ...) +{ + unsigned char *start; + double pts; + + switch (cmd) { + case ADCTRL_RESYNC_STREAM: + case ADCTRL_SKIP_FRAME: + ds_get_packet_pts(sh->ds, &start, &pts); + return CONTROL_TRUE; + } + return CONTROL_UNKNOWN; +} + +static void uninit(sh_audio_t *sh) +{ + struct spdifContext *spdif_ctx = sh->context; + AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx; + + if (lavf_ctx) { + if (lavf_ctx->oformat) + lavf_ctx->oformat->write_trailer(lavf_ctx); + av_freep(&lavf_ctx->pb); + if (lavf_ctx->streams) { + av_freep(&lavf_ctx->streams[0]->codec); + av_freep(&lavf_ctx->streams[0]->info); + av_freep(&lavf_ctx->streams[0]); + } + av_freep(&lavf_ctx->streams); + av_freep(&lavf_ctx->priv_data); + } + av_freep(&lavf_ctx); + av_freep(&spdif_ctx); +} -- cgit v1.2.3