annotate libmpcodecs/ad_ffmpeg.c @ 36963:e539d330c7be

Remove unnecessary bounds checks in Win32 GUI. The checks that the rendered potmeter button doesn't exceed the bounds is not necessary as the item value is already limited within the range of 0 to 100. Patch by Hans-Dieter Kosch, hdkosch kabelbw de.

/*
 * 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"

struct adctx {
    int last_samplerate;
    int srate_changed;
};

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){
        struct adctx *c = lavc_context->opaque;
        c->srate_changed |= c->last_samplerate && c->last_samplerate != samplerate;
        c->last_samplerate = samplerate;
        // 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 && !c->srate_changed)
            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;
    lavc_context->opaque = av_mallocz(sizeof(struct adctx));

    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->channel_layout = sh_audio->channel_layout;
    if (audio_output_channels == 1)
        lavc_context->request_channel_layout = AV_CH_LAYOUT_MONO;
    else if (audio_output_channels == 2)
        lavc_context->request_channel_layout = AV_CH_LAYOUT_STEREO;
    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->opaque);
    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;

#if HAVE_NEON
    if (nb_channels == 2 && bps == 4) {
        const unsigned char *src0 = src[0];
        const unsigned char *src1 = src[1];
        size_t aligned = nb_samples & ~7;
        const unsigned char *src0_end = src0 + aligned*bps;
        while (src0 < src0_end) {
           __asm__ (
               "vld1.32 {q0}, [%0]!\n\t"
               "vld1.32 {q1}, [%1]!\n\t"
               "vld1.32 {q2}, [%0]!\n\t"
               "vld1.32 {q3}, [%1]!\n\t"
               "vst2.32 {q0,q1}, [%2]!\n\t"
               "vst2.32 {q2,q3}, [%2]!\n\t"
               : "+&r"(src0), "+&r"(src1), "+&r"(dst)
               :: "q0", "q1", "q2", "q3", "memory");
        }
        o += aligned*bps;
        nb_samples -= aligned;
    } else if (nb_channels == 2 && bps == 2) {
        const unsigned char *src0 = src[0];
        const unsigned char *src1 = src[1];
        size_t aligned = nb_samples & ~15;
        const unsigned char *src0_end = src0 + aligned*bps;
        while (src0 < src0_end) {
           __asm__ (
               "vld1.16 {q0}, [%0]!\n\t"
               "vld1.16 {q1}, [%1]!\n\t"
               "vld1.16 {q2}, [%0]!\n\t"
               "vld1.16 {q3}, [%1]!\n\t"
               "vst2.16 {q0,q1}, [%2]!\n\t"
               "vst2.16 {q2,q3}, [%2]!\n\t"
               : "+&r"(src0), "+&r"(src1), "+&r"(dst)
               :: "q0", "q1", "q2", "q3", "memory");
        }
        o += aligned*bps;
        nb_samples -= aligned;
    }
#endif
    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 = av_frame_alloc();

    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_frame_free(&frame);
  return len;
}