Mercurial > mplayer.hg
view libmpcodecs/ad_ffmpeg.c @ 36118:2d29160e0957
input: add an option to set the default pausing mode.
author | cigaes |
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date | Fri, 03 May 2013 18:52:54 +0000 |
parents | 2e8a3822bd84 |
children | 3dfc82c0a678 |
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/* * 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. */ #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" #include "dec_audio.h" #include "av_helpers.h" #include "libaf/reorder_ch.h" #include "fmt-conversion.h" static const ad_info_t info = { "FFmpeg/libavcodec audio decoders", "ffmpeg", "Nick Kurshev", "ffmpeg.sf.net", "" }; LIBAD_EXTERN(ffmpeg) #define assert(x) #include "libavcodec/avcodec.h" #include "libavutil/dict.h" static int preinit(sh_audio_t *sh) { sh->audio_out_minsize=AF_NCH*AVCODEC_MAX_AUDIO_FRAME_SIZE; return 1; } static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) { int broken_srate = 0; int samplerate = lavc_context->sample_rate; int sample_format = samplefmt2affmt(av_get_packed_sample_fmt(lavc_context->sample_fmt)); if (!sample_format) sample_format = sh_audio->sample_format; if(sh_audio->wf){ // If the decoder uses the wrong number of channels all is lost anyway. // sh_audio->channels=sh_audio->wf->nChannels; if (lavc_context->codec_id == AV_CODEC_ID_AAC && samplerate == 2*sh_audio->wf->nSamplesPerSec) { broken_srate = 1; } else if (sh_audio->wf->nSamplesPerSec) samplerate=sh_audio->wf->nSamplesPerSec; } if (lavc_context->channels != sh_audio->channels || samplerate != sh_audio->samplerate || sample_format != sh_audio->sample_format) { sh_audio->channels=lavc_context->channels; sh_audio->samplerate=samplerate; sh_audio->sample_format = sample_format; sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; if (broken_srate) mp_msg(MSGT_DECAUDIO, MSGL_WARN, "Ignoring broken container sample rate for AAC with SBR\n"); return 1; } return 0; } static int init(sh_audio_t *sh_audio) { int tries = 0; int x; AVCodecContext *lavc_context; AVCodec *lavc_codec; AVDictionary *opts = NULL; char tmpstr[50]; mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); init_avcodec(); lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); if(!lavc_codec){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); return 0; } lavc_context = avcodec_alloc_context3(lavc_codec); sh_audio->context=lavc_context; snprintf(tmpstr, sizeof(tmpstr), "%f", drc_level); av_dict_set(&opts, "drc_scale", tmpstr, 0); lavc_context->sample_rate = sh_audio->samplerate; lavc_context->bit_rate = sh_audio->i_bps * 8; if(sh_audio->wf){ lavc_context->channels = sh_audio->wf->nChannels; lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; lavc_context->block_align = sh_audio->wf->nBlockAlign; lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; } lavc_context->request_channels = audio_output_channels; lavc_context->codec_tag = sh_audio->format; //FOURCC lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi /* alloc extra data */ if (sh_audio->wf && sh_audio->wf->cbSize > 0) { lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); lavc_context->extradata_size = sh_audio->wf->cbSize; memcpy(lavc_context->extradata, sh_audio->wf + 1, lavc_context->extradata_size); } // for QDM2 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) { lavc_context->extradata = av_malloc(sh_audio->codecdata_len); lavc_context->extradata_size = sh_audio->codecdata_len; memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, lavc_context->extradata_size); } /* open it */ if (avcodec_open2(lavc_context, lavc_codec, &opts) < 0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); return 0; } av_dict_free(&opts); mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); // printf("\nFOURCC: 0x%X\n",sh_audio->format); if(sh_audio->format==0x3343414D){ // MACE 3:1 sh_audio->ds->ss_div = 2*3; // 1 samples/packet sh_audio->ds->ss_mul = sh_audio->wf ? 2*sh_audio->wf->nChannels : 2; // 1 byte*ch/packet } else if(sh_audio->format==0x3643414D){ // MACE 6:1 sh_audio->ds->ss_div = 2*6; // 1 samples/packet sh_audio->ds->ss_mul = sh_audio->wf ? 2*sh_audio->wf->nChannels : 2; // 1 byte*ch/packet } // Decode at least 1 byte: (to get header filled) do { x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); } while (x <= 0 && tries++ < 5); if(x>0) sh_audio->a_buffer_len=x; sh_audio->i_bps=lavc_context->bit_rate/8; if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; switch (lavc_context->sample_fmt) { case AV_SAMPLE_FMT_U8: case AV_SAMPLE_FMT_U8P: case AV_SAMPLE_FMT_S16: case AV_SAMPLE_FMT_S16P: case AV_SAMPLE_FMT_S32: case AV_SAMPLE_FMT_S32P: case AV_SAMPLE_FMT_FLT: case AV_SAMPLE_FMT_FLTP: break; default: return 0; } return 1; } static void uninit(sh_audio_t *sh) { AVCodecContext *lavc_context = sh->context; if (avcodec_close(lavc_context) < 0) mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); av_freep(&lavc_context->extradata); av_freep(&lavc_context); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { AVCodecContext *lavc_context = sh->context; switch(cmd){ case ADCTRL_RESYNC_STREAM: avcodec_flush_buffers(lavc_context); ds_clear_parser(sh->ds); return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static av_always_inline void copy_samples_planar(size_t bps, size_t nb_samples, size_t nb_channels, unsigned char *dst, unsigned char **src) { size_t s, c, o = 0; for (s = 0; s < nb_samples; s++) { for (c = 0; c < nb_channels; c++) { memcpy(dst, src[c] + o, bps); dst += bps; } o += bps; } } static int copy_samples(AVCodecContext *avc, AVFrame *frame, unsigned char *buf, int max_size) { int channels = avc->channels; int sample_size = av_get_bytes_per_sample(avc->sample_fmt); int size = channels * sample_size * frame->nb_samples; if (size > max_size) { av_log(avc, AV_LOG_ERROR, "Buffer overflow while decoding a single frame\n"); return AVERROR(EINVAL); /* same as avcodec_decode_audio3 */ } /* TODO reorder channels at the same time */ if (av_sample_fmt_is_planar(avc->sample_fmt)) { switch (sample_size) { case 1: copy_samples_planar(1, frame->nb_samples, channels, buf, frame->extended_data); break; case 2: copy_samples_planar(2, frame->nb_samples, channels, buf, frame->extended_data); break; case 4: copy_samples_planar(4, frame->nb_samples, channels, buf, frame->extended_data); break; default: copy_samples_planar(sample_size, frame->nb_samples, channels, buf, frame->extended_data); } } else { memcpy(buf, frame->data[0], size); } return size; } static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { unsigned char *start=NULL; int y,len=-1, got_frame; AVFrame *frame = avcodec_alloc_frame(); if (!frame) return AVERROR(ENOMEM); while(len<minlen){ AVPacket pkt; int len2=maxlen; double pts; int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); if(x<=0) { start = NULL; x = 0; ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); if (x <= 0) break; // error } else { int in_size = x; int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); sh_audio->ds->buffer_pos -= in_size - consumed; } av_init_packet(&pkt); pkt.data = start; pkt.size = x; if (pts != MP_NOPTS_VALUE) { sh_audio->pts = pts; sh_audio->pts_bytes = 0; } y=avcodec_decode_audio4(sh_audio->context, frame, &got_frame, &pkt); //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); // LATM may need many packets to find mux info if (y == AVERROR(EAGAIN)) continue; if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } if(!sh_audio->parser && y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) if (!got_frame) continue; len2 = copy_samples(sh_audio->context, frame, buf, maxlen); if (len2 < 0) return len2; if(len2>0){ if (((AVCodecContext *)sh_audio->context)->channels >= 5) { int samplesize = av_get_bytes_per_sample(((AVCodecContext *) sh_audio->context)->sample_fmt); reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, ((AVCodecContext *)sh_audio->context)->channels, len2 / samplesize, samplesize); } //len=len2;break; if(len<0) len=len2; else len+=len2; buf+=len2; maxlen -= len2; sh_audio->pts_bytes += len2; } mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); if (setup_format(sh_audio, sh_audio->context)) break; } av_free(frame); return len; }