Merge remote-tracking branch 'qatar/master'

* qatar/master:
  fate: split off DPCM codec FATE tests into their own file
  fate: split off PCM codec FATE tests into their own file
  libvorbis: K&R reformatting cosmetics
  libmp3lame: K&R formatting cosmetics
  fate: Add a video test for xxan decoder
  mpegvideo_enc: K&R cosmetics (line 1000-2000).
  avconv: K&R cosmetics
  qt-faststart: Fix up indentation
  indeo4: remove two unused variables
  doxygen: cleanup style to support older doxy
  fate: add more tests for VC-1 decoder
  applehttpproto: Apply the same reload interval changes as for the demuxer
  applehttp: Use half the target duration as interval if the playlist didn't update
  applehttp: Use the last segment duration as reload interval
  lagarith: add decode support for arith rgb24 mode

Conflicts:
	avconv.c
	libavcodec/libmp3lame.c
	libavcodec/mpegvideo_enc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 72 insertions, 67 deletions

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index b540362d5c..1b736d0342 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -31,7 +31,7 @@
 #include "mpegaudio.h"
 #include <lame/lame.h>
 
-#define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
+#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
 typedef struct Mp3AudioContext {
     AVClass *class;
     lame_global_flags *gfp;
@@ -62,17 +62,17 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
     lame_set_in_samplerate(s->gfp, avctx->sample_rate);
     lame_set_out_samplerate(s->gfp, avctx->sample_rate);
     lame_set_num_channels(s->gfp, avctx->channels);
-    if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+    if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
         lame_set_quality(s->gfp, 5);
     } else {
         lame_set_quality(s->gfp, avctx->compression_level);
     }
     lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
-    lame_set_brate(s->gfp, avctx->bit_rate/1000);
-    if(avctx->flags & CODEC_FLAG_QSCALE) {
+    lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+    if (avctx->flags & CODEC_FLAG_QSCALE) {
         lame_set_brate(s->gfp, 0);
         lame_set_VBR(s->gfp, vbr_default);
-        lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
+        lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
     }
     lame_set_bWriteVbrTag(s->gfp,0);
 #if FF_API_LAME_GLOBAL_OPTS
@@ -82,14 +82,14 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
     if (lame_init_params(s->gfp) < 0)
         goto err_close;
 
-    avctx->frame_size = lame_get_framesize(s->gfp);
+    avctx->frame_size             = lame_get_framesize(s->gfp);
 
     if(!(avctx->coded_frame= avcodec_alloc_frame())) {
         lame_close(s->gfp);
 
         return AVERROR(ENOMEM);
     }
-    avctx->coded_frame->key_frame= 1;
+    avctx->coded_frame->key_frame = 1;
 
     if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
         int nelem = 2 * avctx->frame_size;
@@ -117,60 +117,62 @@ static const int sSampleRates[] = {
 };
 
 static const int sBitRates[2][3][15] = {
-    {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
-        {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
-        {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
+    {
+        { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
+        { 0, 32, 48, 56, 64,  80,  96,  112, 128, 160, 192, 224, 256, 320, 384 },
+        { 0, 32, 40, 48, 56,  64,  80,  96,  112, 128, 160, 192, 224, 256, 320 }
     },
-    {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
-        {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
-        {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
+    {
+        { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
+        { 0,  8, 16, 24, 32, 40, 48,  56,  64,  80,  96, 112, 128, 144, 160 },
+        { 0,  8, 16, 24, 32, 40, 48,  56,  64,  80,  96, 112, 128, 144, 160 }
     },
 };
 
-static const int sSamplesPerFrame[2][3] =
-{
-    {  384,     1152,    1152 },
-    {  384,     1152,     576 }
+static const int sSamplesPerFrame[2][3] = {
+    { 384, 1152, 1152 },
+    { 384, 1152,  576 }
 };
 
-static const int sBitsPerSlot[3] = {
-    32,
-    8,
-    8
-};
+static const int sBitsPerSlot[3] = { 32, 8, 8 };
 
 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
 {
-    uint32_t header = AV_RB32(data);
-    int layerID = 3 - ((header >> 17) & 0x03);
-    int bitRateID = ((header >> 12) & 0x0f);
+    uint32_t header  = AV_RB32(data);
+    int layerID      = 3 - ((header >> 17) & 0x03);
+    int bitRateID    = ((header >> 12) & 0x0f);
     int sampleRateID = ((header >> 10) & 0x03);
-    int bitsPerSlot = sBitsPerSlot[layerID];
-    int isPadded = ((header >> 9) & 0x01);
-    static int const mode_tab[4]= {2,3,1,0};
-    int mode= mode_tab[(header >> 19) & 0x03];
-    int mpeg_id= mode>0;
+    int bitsPerSlot  = sBitsPerSlot[layerID];
+    int isPadded     = ((header >> 9) & 0x01);
+    static int const mode_tab[4] = { 2, 3, 1, 0 };
+    int mode    = mode_tab[(header >> 19) & 0x03];
+    int mpeg_id = mode > 0;
     int temp0, temp1, bitRate;
 
-    if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
+    if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
+        sampleRateID == 3) {
         return -1;
     }
 
-    if(!samplesPerFrame) samplesPerFrame= &temp0;
-    if(!sampleRate     ) sampleRate     = &temp1;
+    if (!samplesPerFrame)
+        samplesPerFrame = &temp0;
+    if (!sampleRate)
+        sampleRate      = &temp1;
 
-//    *isMono = ((header >>  6) & 0x03) == 0x03;
+    //*isMono = ((header >>  6) & 0x03) == 0x03;
 
-    *sampleRate = sSampleRates[sampleRateID]>>mode;
-    bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
+    *sampleRate      = sSampleRates[sampleRateID] >> mode;
+    bitRate          = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
-//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
+    //av_log(NULL, AV_LOG_DEBUG,
+    //       "sr:%d br:%d spf:%d l:%d m:%d\n",
+    //       *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
 
     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
 }
 
-static int MP3lame_encode_frame(AVCodecContext *avctx,
-                                unsigned char *frame, int buf_size, void *data)
+static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+                                int buf_size, void *data)
 {
     Mp3AudioContext *s = avctx->priv_data;
     int len;
@@ -178,7 +180,7 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
 
     /* lame 3.91 dies on '1-channel interleaved' data */
 
-    if(!data){
+    if (!data){
         lame_result= lame_encode_flush(
                 s->gfp,
                 s->buffer + s->buffer_index,
@@ -237,32 +239,35 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
         }
     }
 
-    if(lame_result < 0){
-        if(lame_result==-1) {
+    if (lame_result < 0) {
+        if (lame_result == -1) {
             /* output buffer too small */
-            av_log(avctx, AV_LOG_ERROR, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s->buffer_index, BUFFER_SIZE - s->buffer_index);
+            av_log(avctx, AV_LOG_ERROR,
+                   "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
+                   s->buffer_index, BUFFER_SIZE - s->buffer_index);
         }
         return -1;
     }
 
     s->buffer_index += lame_result;
 
-    if(s->buffer_index<4)
+    if (s->buffer_index < 4)
         return 0;
 
-    len= mp3len(s->buffer, NULL, NULL);
-//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
-    if(len <= s->buffer_index){
+    len = mp3len(s->buffer, NULL, NULL);
+    //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
+    //       avctx->frame_size, len, s->buffer_index);
+    if (len <= s->buffer_index) {
         memcpy(frame, s->buffer, len);
         s->buffer_index -= len;
 
-        memmove(s->buffer, s->buffer+len, s->buffer_index);
-            //FIXME fix the audio codec API, so we do not need the memcpy()
-/*for(i=0; i<len; i++){
-    av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
-}*/
+        memmove(s->buffer, s->buffer + len, s->buffer_index);
+        // FIXME fix the audio codec API, so we do not need the memcpy()
+        /*for(i=0; i<len; i++) {
+            av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
+        }*/
         return len;
-    }else
+    } else
         return 0;
 }
 
@@ -280,7 +285,7 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
 #define OFFSET(x) offsetof(Mp3AudioContext, x)
 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
 static const AVOption options[] = {
-    { "reservoir",      "Use bit reservoir.",   OFFSET(reservoir),  AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
+    { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
     { NULL },
 };
 
@@ -292,20 +297,20 @@ static const AVClass libmp3lame_class = {
 };
 
 AVCodec ff_libmp3lame_encoder = {
-    .name           = "libmp3lame",
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_MP3,
-    .priv_data_size = sizeof(Mp3AudioContext),
-    .init           = MP3lame_encode_init,
-    .encode         = MP3lame_encode_frame,
-    .close          = MP3lame_encode_close,
-    .capabilities= CODEC_CAP_DELAY,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
+    .name                  = "libmp3lame",
+    .type                  = AVMEDIA_TYPE_AUDIO,
+    .id                    = CODEC_ID_MP3,
+    .priv_data_size        = sizeof(Mp3AudioContext),
+    .init                  = MP3lame_encode_init,
+    .encode                = MP3lame_encode_frame,
+    .close                 = MP3lame_encode_close,
+    .capabilities          = CODEC_CAP_DELAY,
+    .sample_fmts           = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
 #if 2147483647 == INT_MAX
     AV_SAMPLE_FMT_S32,
 #endif
-    AV_SAMPLE_FMT_NONE},
-    .supported_samplerates= sSampleRates,
-    .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
-    .priv_class     = &libmp3lame_class,
+                                                             AV_SAMPLE_FMT_NONE },
+    .supported_samplerates = sSampleRates,
+    .long_name             = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+    .priv_class            = &libmp3lame_class,
 };