cosmetics: ad_ffmpeg.c: reformat

1 files changed, 137 insertions, 131 deletions

diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index fb2d94dbfd..cbd18fddaf 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -20,6 +20,8 @@
 #include <stdlib.h>
 #include <unistd.h>
 
+#include <libavcodec/avcodec.h>
+
 #include "config.h"
 #include "mp_msg.h"
 #include "options.h"
@@ -31,24 +33,19 @@
 
 static const ad_info_t info =
 {
-	"FFmpeg/libavcodec audio decoders",
-	"ffmpeg",
-	"Nick Kurshev",
-	"ffmpeg.sf.net",
-	""
+    "FFmpeg/libavcodec audio decoders",
+    "ffmpeg",
+    "Nick Kurshev",
+    "ffmpeg.sf.net",
+    ""
 };
 
 LIBAD_EXTERN(ffmpeg)
 
-#define assert(x)
-
-#include "libavcodec/avcodec.h"
-
-
 static int preinit(sh_audio_t *sh)
 {
-  sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
-  return 1;
+    sh->audio_out_minsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+    return 1;
 }
 
 /* Prefer playing audio with the samplerate given in container data
@@ -56,16 +53,17 @@ static int preinit(sh_audio_t *sh)
  * from the codec, since if the codec isn't using the correct values for
  * those everything breaks anyway.
  */
-static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
+static int setup_format(sh_audio_t *sh_audio,
+                        const AVCodecContext *lavc_context)
 {
     int sample_format = sh_audio->sample_format;
     switch (lavc_context->sample_fmt) {
-        case SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
-        case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
-        case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
-        case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
-        default:
-            mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
+    case SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
+    case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
+    case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
+    case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
+    default:
+        mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
     }
 
     bool broken_srate        = false;
@@ -82,10 +80,10 @@ static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context
     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->channels = lavc_context->channels;
+        sh_audio->samplerate = samplerate;
         sh_audio->sample_format = sample_format;
-        sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
+        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");
@@ -100,26 +98,28 @@ static int init(sh_audio_t *sh_audio)
     AVCodecContext *lavc_context;
     AVCodec *lavc_codec;
 
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
+    mp_msg(MSGT_DECAUDIO, MSGL_V, "FFmpeg's libavcodec audio codec\n");
 
     lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
-    if(!lavc_codec){
-	mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
-	return 0;
+    if (!lavc_codec) {
+        mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
+                "Cannot find codec '%s' in libavcodec...\n",
+                sh_audio->codec->dll);
+        return 0;
     }
 
     lavc_context = avcodec_alloc_context();
-    sh_audio->context=lavc_context;
+    sh_audio->context = lavc_context;
 
     lavc_context->drc_scale = opts->drc_level;
     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;
+    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 = opts->audio_output_channels;
     lavc_context->codec_tag = sh_audio->format; //FOURCC
@@ -135,8 +135,8 @@ static int init(sh_audio_t *sh_audio)
     }
 
     // for QDM2
-    if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
-    {
+    if (sh_audio->codecdata_len && sh_audio->codecdata &&
+            !lavc_context->extradata) {
         lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
                                             FF_INPUT_BUFFER_PADDING_SIZE);
         lavc_context->extradata_size = sh_audio->codecdata_len;
@@ -146,52 +146,51 @@ static int init(sh_audio_t *sh_audio)
 
     /* open it */
     if (avcodec_open(lavc_context, lavc_codec) < 0) {
-        mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
+        mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
         return 0;
     }
-   mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
+    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 = 2*sh_audio->wf->nChannels; // 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 = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
-   }
+    if (sh_audio->format == 0x3343414D) {
+        // MACE 3:1
+        sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
+        sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 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 = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
+    }
 
-   // Decode at least 1 byte:  (to get header filled)
-   for (int tries = 0;;) {
-       int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
-                            sh_audio->a_buffer_size);
-       if (x > 0) {
-           sh_audio->a_buffer_len = x;
-           break;
-       }
-       if (++tries >= 5) {
-           mp_msg(MSGT_DECAUDIO, MSGL_ERR,
-                  "ad_ffmpeg: initial decode failed\n");
-           return 0;
-       }
-   }
+    // Decode at least 1 byte:  (to get header filled)
+    for (int tries = 0;;) {
+        int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
+                             sh_audio->a_buffer_size);
+        if (x > 0) {
+            sh_audio->a_buffer_len = x;
+            break;
+        }
+        if (++tries >= 5) {
+            mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+                   "ad_ffmpeg: initial decode failed\n");
+            return 0;
+        }
+    }
 
-  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;
+    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 SAMPLE_FMT_U8:
-      case SAMPLE_FMT_S16:
-      case SAMPLE_FMT_S32:
-      case SAMPLE_FMT_FLT:
-          break;
-      default:
-          return 0;
-  }
-  return 1;
+    switch (lavc_context->sample_fmt) {
+    case SAMPLE_FMT_U8:
+    case SAMPLE_FMT_S16:
+    case SAMPLE_FMT_S32:
+    case SAMPLE_FMT_FLT:
+        break;
+    default:
+        return 0;
+    }
+    return 1;
 }
 
 static void uninit(sh_audio_t *sh)
@@ -199,77 +198,84 @@ static void uninit(sh_audio_t *sh)
     AVCodecContext *lavc_context = sh->context;
 
     if (avcodec_close(lavc_context) < 0)
-	mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
+        mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
     av_freep(&lavc_context->extradata);
     av_freep(&lavc_context);
 }
 
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
 {
     AVCodecContext *lavc_context = sh->context;
-    switch(cmd){
+    switch (cmd) {
     case ADCTRL_RESYNC_STREAM:
         avcodec_flush_buffers(lavc_context);
         ds_clear_parser(sh->ds);
-    return CONTROL_TRUE;
+        return CONTROL_TRUE;
     }
     return CONTROL_UNKNOWN;
 }
 
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
+                        int maxlen)
 {
-    unsigned char *start=NULL;
-    int y,len=-1;
-    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_audio3(sh_audio->context,(int16_t*)buf,&len2,&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(len2>0){
-	  if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
-            int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
-                                    sh_audio->context)->sample_fmt) / 8;
-            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);
+    AVCodecContext *avctx = sh_audio->context;
+
+    unsigned char *start = NULL;
+    int y, len = -1;
+    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_audio3(avctx, (int16_t *)buf, &len2, &pkt);
+        // 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 (len2 > 0) {
+            if (avctx->channels >= 5) {
+                int samplesize = av_get_bits_per_sample_format(
+                                                    avctx->sample_fmt) / 8;
+                reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
+                                    AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+                                    avctx->channels,
+                                    len2 / samplesize, samplesize);
+            }
+            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))
+        if (setup_format(sh_audio, avctx))
             break;
     }
-  return len;
+    return len;
 }