Merge remote-tracking branch 'qatar/master'

* qatar/master:
  Remove ffmpeg.
  aacenc: Simplify windowing
  aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
  aacenc: Deinterleave input samples before processing.
  aacenc: Store channel count in AACEncContext.
  aacenc: Move Q^3/4 calculation to it's own table
  aacenc: Request normalized float samples instead of converting s16 samples to float.
  aacpsy: Replace an if with FFMAX in LAME windowing.
  aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
  aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
  aacenc: cosmetics: move init() and end() to the bottom of the file.
  aacenc: aac_encode_init() cleanup
  XWD encoder and decoder
  vc1: don't read the interpfrm and bfraction elements for interlaced frames
  mxfdec: fix memleak on mxf_read_close()
  westwood: split the AUD and VQA demuxers into separate files.

Conflicts:
	.gitignore
	Changelog
	Makefile
	configure
	doc/ffmpeg.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/aacenc.c
	libavcodec/allcodecs.c
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/Makefile
	libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -15,6 +15,7 @@ version next:
 - amerge audio filter
 - GSM audio parser
 - SMJPEG muxer
+- XWD encoder and decoder
 - Automatic thread count based on detection number of (available) CPU cores
 - y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
 - ffprobe -show_error option

doc/general.texi

@@ -399,6 +399,8 @@ following image formats are supported:
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
+@item XWD  @tab X @tab X
+    @tab X Window Dump image format
 @end multitable
 
 @code{X} means that encoding (resp. decoding) is supported.

libavcodec/Makefile

@@ -470,6 +470,8 @@ OBJS-$(CONFIG_XBIN_DECODER)            += bintext.o cga_data.o
 OBJS-$(CONFIG_XL_DECODER)              += xl.o
 OBJS-$(CONFIG_XSUB_DECODER)            += xsubdec.o
 OBJS-$(CONFIG_XSUB_ENCODER)            += xsubenc.o
+OBJS-$(CONFIG_XWD_DECODER)             += xwddec.o
+OBJS-$(CONFIG_XWD_ENCODER)             += xwdenc.o
 OBJS-$(CONFIG_Y41P_DECODER)            += y41pdec.o
 OBJS-$(CONFIG_Y41P_ENCODER)            += y41penc.o
 OBJS-$(CONFIG_YOP_DECODER)             += yop.o

libavcodec/aaccoder.c

@@ -110,14 +110,15 @@ static av_always_inline float quantize_and_encode_band_cost_template(
                                 int *bits, int BT_ZERO, int BT_UNSIGNED,
                                 int BT_PAIR, int BT_ESC)
 {
-    const float IQ = ff_aac_pow2sf_tab[POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
-    const float  Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
+    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
+    const float Q   = ff_aac_pow2sf_tab [q_idx];
+    const float Q34 = ff_aac_pow34sf_tab[q_idx];
+    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
     const float CLIPPED_ESCAPE = 165140.0f*IQ;
     int i, j;
     float cost = 0;
     const int dim = BT_PAIR ? 2 : 4;
     int resbits = 0;
-    const float  Q34 = sqrtf(Q * sqrtf(Q));
     const int range  = aac_cb_range[cb];
     const int maxval = aac_cb_maxval[cb];
     int off;
@@ -420,7 +421,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     const int run_esc  = (1 << run_bits) - 1;
     int idx, ppos, count;
     int stackrun[120], stackcb[120], stack_len;
-    float next_minrd = INFINITY;
+    float next_minbits = INFINITY;
     int next_mincb = 0;
 
     abs_pow34_v(s->scoefs, sce->coeffs, 1024);
@@ -434,7 +435,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
         size = sce->ics.swb_sizes[swb];
         if (sce->zeroes[win*16 + swb]) {
             float cost_stay_here = path[swb][0].cost;
-            float cost_get_here  = next_minrd + run_bits + 4;
+            float cost_get_here  = next_minbits + run_bits + 4;
             if (   run_value_bits[sce->ics.num_windows == 8][path[swb][0].run]
                 != run_value_bits[sce->ics.num_windows == 8][path[swb][0].run+1])
                 cost_stay_here += run_bits;
@@ -447,7 +448,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                 path[swb+1][0].cost     = cost_stay_here;
                 path[swb+1][0].run      = path[swb][0].run + 1;
             }
-            next_minrd = path[swb+1][0].cost;
+            next_minbits = path[swb+1][0].cost;
             next_mincb = 0;
             for (cb = 1; cb < 12; cb++) {
                 path[swb+1][cb].cost = 61450;
@@ -455,10 +456,10 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                 path[swb+1][cb].run = 0;
             }
         } else {
-            float minrd = next_minrd;
+            float minbits = next_minbits;
             int mincb = next_mincb;
             int startcb = sce->band_type[win*16+swb];
-            next_minrd = INFINITY;
+            next_minbits = INFINITY;
             next_mincb = 0;
             for (cb = 0; cb < startcb; cb++) {
                 path[swb+1][cb].cost = 61450;
@@ -467,15 +468,15 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             }
             for (cb = startcb; cb < 12; cb++) {
                 float cost_stay_here, cost_get_here;
-                float rd = 0.0f;
+                float bits = 0.0f;
                 for (w = 0; w < group_len; w++) {
-                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
-                                             s->scoefs + start + w*128, size,
-                                             sce->sf_idx[(win+w)*16+swb], cb,
-                                             0, INFINITY, NULL);
+                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                               s->scoefs + start + w*128, size,
+                                               sce->sf_idx[(win+w)*16+swb], cb,
+                                               0, INFINITY, NULL);
                 }
-                cost_stay_here = path[swb][cb].cost + rd;
-                cost_get_here  = minrd              + rd + run_bits + 4;
+                cost_stay_here = path[swb][cb].cost + bits;
+                cost_get_here  = minbits            + bits + run_bits + 4;
                 if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
                     != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
                     cost_stay_here += run_bits;
@@ -488,8 +489,8 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                     path[swb+1][cb].cost     = cost_stay_here;
                     path[swb+1][cb].run      = path[swb][cb].run + 1;
                 }
-                if (path[swb+1][cb].cost < next_minrd) {
-                    next_minrd = path[swb+1][cb].cost;
+                if (path[swb+1][cb].cost < next_minbits) {
+                    next_minbits = path[swb+1][cb].cost;
                     next_mincb = cb;
                 }
             }

libavcodec/aacenc.c

@@ -46,6 +46,14 @@
 
 #define AAC_MAX_CHANNELS 6
 
+#define ERROR_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
+        return AVERROR(EINVAL); \
+    }
+
+float ff_aac_pow34sf_tab[428];
+
 static const uint8_t swb_size_1024_96[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
@@ -135,6 +143,18 @@ static const uint8_t aac_chan_configs[6][5] = {
  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
 };
 
+/**
+ * Table to remap channels from Libav's default order to AAC order.
+ */
+static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+    { 0 },
+    { 0, 1 },
+    { 2, 0, 1 },
+    { 2, 0, 1, 3 },
+    { 2, 0, 1, 3, 4 },
+    { 2, 0, 1, 4, 5, 3 },
+};
+
 /**
  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -147,7 +167,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
     init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
     put_bits(&pb, 5, 2); //object type - AAC-LC
     put_bits(&pb, 4, s->samplerate_index); //sample rate index
-    put_bits(&pb, 4, avctx->channels);
+    put_bits(&pb, 4, s->channels);
     //GASpecificConfig
     put_bits(&pb, 1, 0); //frame length - 1024 samples
     put_bits(&pb, 1, 0); //does not depend on core coder
@@ -160,117 +180,80 @@ static void put_audio_specific_config(AVCodecContext *avctx)
     flush_put_bits(&pb);
 }
 
-static av_cold int aac_encode_init(AVCodecContext *avctx)
-{
-    AACEncContext *s = avctx->priv_data;
-    int i;
-    const uint8_t *sizes[2];
-    uint8_t grouping[AAC_MAX_CHANNELS];
-    int lengths[2];
-
-    avctx->frame_size = 1024;
+#define WINDOW_FUNC(type) \
+static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio)
 
-    for (i = 0; i < 16; i++)
-        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
-            break;
-    if (i == 16) {
-        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
-        return -1;
-    }
-    if (avctx->channels > AAC_MAX_CHANNELS) {
-        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
-        return -1;
-    }
-    if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
-        av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
-        return -1;
-    }
-    if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
-        av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
-        return -1;
-    }
-    s->samplerate_index = i;
-
-    dsputil_init(&s->dsp, avctx);
-    ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
-    ff_mdct_init(&s->mdct128,   8, 0, 1.0);
-    // window init
-    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
-    ff_init_ff_sine_windows(10);
-    ff_init_ff_sine_windows(7);
-
-    s->chan_map           = aac_chan_configs[avctx->channels-1];
-    s->samples            = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
-    s->cpe                = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
-    avctx->extradata      = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
-    avctx->extradata_size = 5;
-    put_audio_specific_config(avctx);
+WINDOW_FUNC(only_long)
+{
+    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+    float *out = sce->ret;
 
-    sizes[0]   = swb_size_1024[i];
-    sizes[1]   = swb_size_128[i];
-    lengths[0] = ff_aac_num_swb_1024[i];
-    lengths[1] = ff_aac_num_swb_128[i];
-    for (i = 0; i < s->chan_map[0]; i++)
-        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
-    ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
-    s->psypp = ff_psy_preprocess_init(avctx);
-    s->coder = &ff_aac_coders[s->options.aac_coder];
+    dsp->vector_fmul        (out,        audio,        lwindow, 1024);
+    dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
+}
 
-    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+WINDOW_FUNC(long_start)
+{
+    const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+    const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+    float *out = sce->ret;
+
+    dsp->vector_fmul(out, audio, lwindow, 1024);
+    memcpy(out + 1024, audio, sizeof(out[0]) * 448);
+    dsp->vector_fmul_reverse(out + 1024 + 448, audio, swindow, 128);
+    memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
+}
 
-    ff_aac_tableinit();
+WINDOW_FUNC(long_stop)
+{
+    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+    const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+    float *out = sce->ret;
+
+    memset(out, 0, sizeof(out[0]) * 448);
+    dsp->vector_fmul(out + 448, audio + 448, swindow, 128);
+    memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
+    dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
+}
 
-    return 0;
+WINDOW_FUNC(eight_short)
+{
+    const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+    const float *in = audio + 448;
+    float *out = sce->ret;
+
+    for (int w = 0; w < 8; w++) {
+        dsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
+        out += 128;
+        in  += 128;
+        dsp->vector_fmul_reverse(out, in, swindow, 128);
+        out += 128;
+    }
 }
 
-static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
-                                  SingleChannelElement *sce, short *audio)
+static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = {
+    [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
+    [LONG_START_SEQUENCE]  = apply_long_start_window,
+    [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
+    [LONG_STOP_SEQUENCE]   = apply_long_stop_window
+};
+
+static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
+                                  float *audio)
 {
-    int i, k;
-    const int chans = avctx->channels;
-    const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
-    const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+    int i;
     float *output = sce->ret;
 
-    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
-        memcpy(output, sce->saved, sizeof(float)*1024);
-        if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
-            memset(output, 0, sizeof(output[0]) * 448);
-            for (i = 448; i < 576; i++)
-                output[i] = sce->saved[i] * pwindow[i - 448];
-            for (i = 576; i < 704; i++)
-                output[i] = sce->saved[i];
-        }
-        if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
-            for (i = 0; i < 1024; i++) {
-                output[i+1024]         = audio[i * chans] * lwindow[1024 - i - 1];
-                sce->saved[i] = audio[i * chans] * lwindow[i];
-            }
-        } else {
-            for (i = 0; i < 448; i++)
-                output[i+1024]         = audio[i * chans];
-            for (; i < 576; i++)
-                output[i+1024]         = audio[i * chans] * swindow[576 - i - 1];
-            memset(output+1024+576, 0, sizeof(output[0]) * 448);
-            for (i = 0; i < 1024; i++)
-                sce->saved[i] = audio[i * chans];
-        }
+    apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio);
+
+    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
-    } else {
-        for (k = 0; k < 1024; k += 128) {
-            for (i = 448 + k; i < 448 + k + 256; i++)
-                output[i - 448 - k] = (i < 1024)
-                                         ? sce->saved[i]
-                                         : audio[(i-1024)*chans];
-            s->dsp.vector_fmul        (output,     output, k ?  swindow : pwindow, 128);
-            s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
-            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
-        }
-        for (i = 0; i < 1024; i++)
-            sce->saved[i] = audio[i * chans];
-    }
+    else
+        for (i = 0; i < 1024; i += 128)
+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
+    memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
 }
 
 /**
@@ -488,11 +471,37 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
     put_bits(&s->pb, 12 - padbits, 0);
 }
 
+/*
+ * Deinterleave input samples.
+ * Channels are reordered from Libav's default order to AAC order.
+ */
+static void deinterleave_input_samples(AACEncContext *s,
+                                       const float *samples)
+{
+    int ch, i;
+    const int sinc = s->channels;
+    const uint8_t *channel_map = aac_chan_maps[sinc - 1];
+
+    /* deinterleave and remap input samples */
+    for (ch = 0; ch < sinc; ch++) {
+        const float *sptr = samples + channel_map[ch];
+
+        /* copy last 1024 samples of previous frame to the start of the current frame */
+        memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][1024], 1024 * sizeof(s->planar_samples[0][0]));
+
+        /* deinterleave */
+        for (i = 1024; i < 1024 * 2; i++) {
+            s->planar_samples[ch][i] = *sptr;
+            sptr += sinc;
+        }
+    }
+}
+
 static int aac_encode_frame(AVCodecContext *avctx,
                             uint8_t *frame, int buf_size, void *data)
 {
     AACEncContext *s = avctx->priv_data;
-    int16_t *samples = s->samples, *samples2, *la;
+    float **samples = s->planar_samples, *samples2, *la, *overlap;
     ChannelElement *cpe;
     int i, ch, w, g, chans, tag, start_ch;
     int chan_el_counter[4];
@@ -500,27 +509,15 @@ static int aac_encode_frame(AVCodecContext *avctx,
 
     if (s->last_frame)
         return 0;
+
     if (data) {
-        if (!s->psypp) {
-            memcpy(s->samples + 1024 * avctx->channels, data,
-                   1024 * avctx->channels * sizeof(s->samples[0]));
-        } else {
-            start_ch = 0;
-            samples2 = s->samples + 1024 * avctx->channels;
-            for (i = 0; i < s->chan_map[0]; i++) {
-                tag = s->chan_map[i+1];
-                chans = tag == TYPE_CPE ? 2 : 1;
-                ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
-                                  samples2 + start_ch, start_ch, chans);
-                start_ch += chans;
-            }
-        }
+        deinterleave_input_samples(s, data);
+        if (s->psypp)
+            ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
     }
-    if (!avctx->frame_number) {
-        memcpy(s->samples, s->samples + 1024 * avctx->channels,
-               1024 * avctx->channels * sizeof(s->samples[0]));
+
+    if (!avctx->frame_number)
         return 0;
-    }
 
     start_ch = 0;
     for (i = 0; i < s->chan_map[0]; i++) {
@@ -531,8 +528,9 @@ static int aac_encode_frame(AVCodecContext *avctx,
         for (ch = 0; ch < chans; ch++) {
             IndividualChannelStream *ics = &cpe->ch[ch].ics;
             int cur_channel = start_ch + ch;
-            samples2 = samples + cur_channel;
-            la       = samples2 + (448+64) * avctx->channels;
+            overlap  = &samples[cur_channel][0];
+            samples2 = overlap + 1024;
+            la       = samples2 + (448+64);
             if (!data)
                 la = NULL;
             if (tag == TYPE_LFE) {
@@ -560,7 +558,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];
 
-            apply_window_and_mdct(avctx, s, &cpe->ch[ch], samples2);
+            apply_window_and_mdct(s, &cpe->ch[ch], overlap);
         }
         start_ch += chans;
     }
@@ -626,8 +624,8 @@ static int aac_encode_frame(AVCodecContext *avctx,
         }
 
         frame_bits = put_bits_count(&s->pb);
-        if (frame_bits <= 6144 * avctx->channels - 3) {
-            s->psy.bitres.bits = frame_bits / avctx->channels;
+        if (frame_bits <= 6144 * s->channels - 3) {
+            s->psy.bitres.bits = frame_bits / s->channels;
             break;
         }
 
@@ -648,8 +646,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
 
     if (!data)
         s->last_frame = 1;
-    memcpy(s->samples, s->samples + 1024 * avctx->channels,
-           1024 * avctx->channels * sizeof(s->samples[0]));
+
     return put_bits_count(&s->pb)>>3;
 }
 
@@ -660,12 +657,109 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);
     ff_psy_end(&s->psy);
-    ff_psy_preprocess_end(s->psypp);
-    av_freep(&s->samples);
+    if (s->psypp)
+        ff_psy_preprocess_end(s->psypp);
+    av_freep(&s->buffer.samples);
     av_freep(&s->cpe);
     return 0;
 }
 
+static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
+{
+    int ret = 0;
+
+    dsputil_init(&s->dsp, avctx);
+
+    // window init
+    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
+    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
+    ff_init_ff_sine_windows(10);
+    ff_init_ff_sine_windows(7);
+
+    if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
+        return ret;
+    if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))
+        return ret;
+
+    return 0;
+}
+
+static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
+{
+    FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
+    FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
+    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+
+    for(int ch = 0; ch < s->channels; ch++)
+        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
+
+    return 0;
+alloc_fail:
+    return AVERROR(ENOMEM);
+}
+
+static av_cold int aac_encode_init(AVCodecContext *avctx)
+{
+    AACEncContext *s = avctx->priv_data;
+    int i, ret = 0;
+    const uint8_t *sizes[2];
+    uint8_t grouping[AAC_MAX_CHANNELS];
+    int lengths[2];
+
+    avctx->frame_size = 1024;
+
+    for (i = 0; i < 16; i++)
+        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
+            break;
+
+    s->channels = avctx->channels;
+
+    ERROR_IF(i == 16,
+             "Unsupported sample rate %d\n", avctx->sample_rate);
+    ERROR_IF(s->channels > AAC_MAX_CHANNELS,
+             "Unsupported number of channels: %d\n", s->channels);
+    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
+             "Unsupported profile %d\n", avctx->profile);
+    ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
+             "Too many bits per frame requested\n");
+
+    s->samplerate_index = i;
+
+    s->chan_map = aac_chan_configs[s->channels-1];
+
+    if (ret = dsp_init(avctx, s))
+        goto fail;
+
+    if (ret = alloc_buffers(avctx, s))
+        goto fail;
+
+    avctx->extradata_size = 5;
+    put_audio_specific_config(avctx);
+
+    sizes[0]   = swb_size_1024[i];
+    sizes[1]   = swb_size_128[i];
+    lengths[0] = ff_aac_num_swb_1024[i];
+    lengths[1] = ff_aac_num_swb_128[i];
+    for (i = 0; i < s->chan_map[0]; i++)
+        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+    if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
+        goto fail;
+    s->psypp = ff_psy_preprocess_init(avctx);
+    s->coder = &ff_aac_coders[s->options.aac_coder];
+
+    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+
+    ff_aac_tableinit();
+
+    for (i = 0; i < 428; i++)
+        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
+
+    return 0;
+fail:
+    aac_encode_end(avctx);
+    return ret;
+}
+
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
     {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
@@ -692,7 +786,7 @@ AVCodec ff_aac_encoder = {
     .encode         = aac_encode_frame,
     .close          = aac_encode_end,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
     .priv_class = &aacenc_class,
 };

libavcodec/aacenc.h

@@ -61,9 +61,10 @@ typedef struct AACEncContext {
     FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
     FFTContext mdct128;                          ///< short (128 samples) frame transform context
     DSPContext  dsp;
-    int16_t *samples;                            ///< saved preprocessed input
+    float *planar_samples[6];                    ///< saved preprocessed input
 
     int samplerate_index;                        ///< MPEG-4 samplerate index
+    int channels;                                ///< channel count
     const uint8_t *chan_map;                     ///< channel configuration map
 
     ChannelElement *cpe;                         ///< channel elements
@@ -75,6 +76,12 @@ typedef struct AACEncContext {
     float lambda;
     DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
     DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients
+
+    struct {
+        float *samples;
+    } buffer;
 } AACEncContext;
 
+extern float ff_aac_pow34sf_tab[428];
+
 #endif /* AVCODEC_AACENC_H */

libavcodec/aacpsy.c

@@ -400,7 +400,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
         int stay_short = 0;
         for (i = 0; i < 8; i++) {
             for (j = 0; j < 128; j++) {
-                v = iir_filter(la[(i*128+j)*ctx->avctx->channels], pch->iir_state);
+                v = iir_filter(la[i*128+j], pch->iir_state);
                 sum += v*v;
             }
             s[i]  = sum;
@@ -776,9 +776,8 @@ static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int u
     ctx->next_window_seq = blocktype;
 }
 
-static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
-                                       const int16_t *audio, const int16_t *la,
-                                       int channel, int prev_type)
+static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
+                                       const float *la, int channel, int prev_type)
 {
     AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
     AacPsyChannel *pch  = &pctx->ch[channel];
@@ -795,20 +794,20 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
         float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
         float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
         float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
-        int chans = ctx->avctx->channels;
-        const int16_t *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN) * chans;
+        const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
         int j, att_sum = 0;
 
         /* LAME comment: apply high pass filter of fs/4 */
         for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
             float sum1, sum2;
-            sum1 = firbuf[(i + ((PSY_LAME_FIR_LEN - 1) / 2)) * chans];
+            sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
             sum2 = 0.0;
             for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
-                sum1 += psy_fir_coeffs[j] * (firbuf[(i + j) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j) * chans]);
-                sum2 += psy_fir_coeffs[j + 1] * (firbuf[(i + j + 1) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j - 1) * chans]);
+                sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
+                sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
             }
-            hpfsmpl[i] = sum1 + sum2;
+            /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768. Tuning this for normalized floats would be difficult. */
+            hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
         }
 
         /* Calculate the energies of each sub-shortblock */
@@ -823,16 +822,15 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
             float const *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS);
             float p = 1.0f;
             for (; pf < pfe; pf++)
-                if (p < fabsf(*pf))
-                    p = fabsf(*pf);
+                p = FFMAX(p, fabsf(*pf));
             pch->prev_energy_subshort[i] = energy_subshort[i + PSY_LAME_NUM_SUBBLOCKS] = p;
             energy_short[1 + i / PSY_LAME_NUM_SUBBLOCKS] += p;
-            /* FIXME: The indexes below are [i + 3 - 2] in the LAME source.
-             *          Obviously the 3 and 2 have some significance, or this would be just [i + 1]
-             *          (which is what we use here). What the 3 stands for is ambigious, as it is both
-             *          number of short blocks, and the number of sub-short blocks.
-             *          It seems that LAME is comparing each sub-block to sub-block + 1 in the
-             *          previous block.
+            /* NOTE: The indexes below are [i + 3 - 2] in the LAME source.
+             *       Obviously the 3 and 2 have some significance, or this would be just [i + 1]
+             *       (which is what we use here). What the 3 stands for is ambiguous, as it is both
+             *       number of short blocks, and the number of sub-short blocks.
+             *       It seems that LAME is comparing each sub-block to sub-block + 1 in the
+             *       previous block.
              */
             if (p > energy_subshort[i + 1])
                 p = p / energy_subshort[i + 1];

libavcodec/allcodecs.c

@@ -245,6 +245,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER (XAN_WC3, xan_wc3);
     REGISTER_DECODER (XAN_WC4, xan_wc4);
     REGISTER_DECODER (XL, xl);
+    REGISTER_ENCDEC  (XWD, xwd);
     REGISTER_ENCDEC  (Y41P, y41p);
     REGISTER_DECODER (YOP, yop);
     REGISTER_ENCDEC  (YUV4, yuv4);

libavcodec/avcodec.h

@@ -255,6 +255,7 @@ enum CodecID {
     CODEC_ID_VBLE,
     CODEC_ID_DXTORY,
     CODEC_ID_V410,
+    CODEC_ID_XWD,
     CODEC_ID_Y41P       = MKBETAG('Y','4','1','P'),
     CODEC_ID_UTVIDEO = 0x800,
     CODEC_ID_ESCAPE130  = MKBETAG('E','1','3','0'),

libavcodec/psymodel.c

@@ -112,20 +112,15 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
     return ctx;
 }
 
-void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
-                       const int16_t *audio, int16_t *dest,
-                       int tag, int channels)
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
 {
-    int ch, i;
+    int ch;
+    int frame_size = ctx->avctx->frame_size;
+
     if (ctx->fstate) {
         for (ch = 0; ch < channels; ch++)
-            ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
-                          audio + ch, ctx->avctx->channels,
-                          dest  + ch, ctx->avctx->channels);
-    } else {
-        for (ch = 0; ch < channels; ch++)
-            for (i = 0; i < ctx->avctx->frame_size; i++)
-                dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
+            ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
+                              &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
     }
 }
 

libavcodec/psymodel.h

@@ -109,7 +109,7 @@ typedef struct FFPsyModel {
      *
      * @return suggested window information in a structure
      */
-    FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
+    FFPsyWindowInfo (*window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type);
 
     /**
      * Perform psychoacoustic analysis and set band info (threshold, energy) for a group of channels.
@@ -174,14 +174,10 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
  * Preprocess several channel in audio frame in order to compress it better.
  *
  * @param ctx      preprocessing context
- * @param audio    samples to preprocess
- * @param dest     place to put filtered samples
- * @param tag      channel number
- * @param channels number of channel to preprocess (some additional work may be done on stereo pair)
+ * @param audio    samples to be filtered (in place)
+ * @param channels number of channel to preprocess
  */
-void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
-                       const int16_t *audio, int16_t *dest,
-                       int tag, int channels);
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels);
 
 /**
  * Cleanup audio preprocessing module.

libavcodec/vc1.c

@@ -918,13 +918,15 @@ int vc1_parse_frame_header_adv(VC1Context *v, GetBitContext* gb)
         }
         goto parse_common_info;
     }
-    if (v->finterpflag)
-        v->interpfrm = get_bits1(gb);
-    if (v->s.pict_type == AV_PICTURE_TYPE_B) {
-        v->bfraction_lut_index = get_vlc2(gb, ff_vc1_bfraction_vlc.table, VC1_BFRACTION_VLC_BITS, 1);
-        v->bfraction           = ff_vc1_bfraction_lut[v->bfraction_lut_index];
-        if (v->bfraction == 0) {
-            v->s.pict_type = AV_PICTURE_TYPE_BI; /* XXX: should not happen here */
+    if (v->fcm == PROGRESSIVE) {
+        if (v->finterpflag)
+            v->interpfrm = get_bits1(gb);
+        if (v->s.pict_type == AV_PICTURE_TYPE_B) {
+            v->bfraction_lut_index = get_vlc2(gb, ff_vc1_bfraction_vlc.table, VC1_BFRACTION_VLC_BITS, 1);
+            v->bfraction           = ff_vc1_bfraction_lut[v->bfraction_lut_index];
+            if (v->bfraction == 0) {
+                v->s.pict_type = AV_PICTURE_TYPE_BI; /* XXX: should not happen here */
+            }
         }
     }
 

libavcodec/version.h

@@ -21,8 +21,8 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 53
-#define LIBAVCODEC_VERSION_MINOR 57
-#define LIBAVCODEC_VERSION_MICRO 105
+#define LIBAVCODEC_VERSION_MINOR 58
+#define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \

libavcodec/xwd.h

+/*
+ * XWD image format
+ *
+ * Copyright (c) 2012 Paul B Mahol
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_XWD_H
+#define AVCODEC_XWD_H
+
+#define XWD_VERSION         7
+#define XWD_HEADER_SIZE     100
+#define XWD_CMAP_SIZE       12
+
+#define XWD_XY_BITMAP       0
+#define XWD_XY_PIXMAP       1
+#define XWD_Z_PIXMAP        2
+
+#define XWD_STATIC_GRAY     0
+#define XWD_GRAY_SCALE      1
+#define XWD_STATIC_COLOR    2
+#define XWD_PSEUDO_COLOR    3
+#define XWD_TRUE_COLOR      4
+#define XWD_DIRECT_COLOR    5
+
+#endif /* AVCODEC_XWD_H */

libavcodec/xwddec.c

+/*
+ * XWD image format
+ *
+ * Copyright (c) 2012 Paul B Mahol
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/imgutils.h"
+#include "avcodec.h"
+#include "bytestream.h"
+#include "xwd.h"
+
+static av_cold int xwd_decode_init(AVCodecContext *avctx)
+{
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int xwd_decode_frame(AVCodecContext *avctx, void *data,
+                            int *data_size, AVPacket *avpkt)
+{
+    AVFrame *p = avctx->coded_frame;
+    const uint8_t *buf = avpkt->data;
+    int i, ret, buf_size = avpkt->size;
+    uint32_t version, header_size, vclass, ncolors;
+    uint32_t xoffset, be, bpp, lsize, rsize;
+    uint32_t pixformat, pixdepth, bunit, bitorder, bpad;
+    uint32_t rgb[3];
+    uint8_t *ptr;
+
+    if (buf_size < XWD_HEADER_SIZE)
+        return AVERROR_INVALIDDATA;
+
+    header_size = bytestream_get_be32(&buf);
+    if (buf_size < header_size)
+        return AVERROR_INVALIDDATA;
+
+    version = bytestream_get_be32(&buf);
+    if (version != XWD_VERSION) {
+        av_log(avctx, AV_LOG_ERROR, "unsupported version\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (header_size < XWD_HEADER_SIZE) {
+        av_log(avctx, AV_LOG_ERROR, "invalid header size\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    pixformat     = bytestream_get_be32(&buf);
+    pixdepth      = bytestream_get_be32(&buf);
+    avctx->width  = bytestream_get_be32(&buf);
+    avctx->height = bytestream_get_be32(&buf);
+    xoffset       = bytestream_get_be32(&buf);
+    be            = bytestream_get_be32(&buf);
+    bunit         = bytestream_get_be32(&buf);
+    bitorder      = bytestream_get_be32(&buf);
+    bpad          = bytestream_get_be32(&buf);
+    bpp           = bytestream_get_be32(&buf);
+    lsize         = bytestream_get_be32(&buf);
+    vclass        = bytestream_get_be32(&buf);
+    rgb[0]        = bytestream_get_be32(&buf);
+    rgb[1]        = bytestream_get_be32(&buf);
+    rgb[2]        = bytestream_get_be32(&buf);
+    buf          += 8;
+    ncolors       = bytestream_get_be32(&buf);
+    buf          += header_size - (XWD_HEADER_SIZE - 20);
+
+    av_log(avctx, AV_LOG_DEBUG, "pixformat %d, pixdepth %d, bunit %d, bitorder %d, bpad %d\n",
+           pixformat, pixdepth, bunit, bitorder, bpad);
+    av_log(avctx, AV_LOG_DEBUG, "vclass %d, ncolors %d, bpp %d, be %d, lsize %d, xoffset %d\n",
+           vclass, ncolors, bpp, be, lsize, xoffset);
+    av_log(avctx, AV_LOG_DEBUG, "red %0x, green %0x, blue %0x\n", rgb[0], rgb[1], rgb[2]);
+
+    if (pixformat > XWD_Z_PIXMAP) {
+        av_log(avctx, AV_LOG_ERROR, "invalid pixmap format\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (pixdepth == 0 || pixdepth > 32) {
+        av_log(avctx, AV_LOG_ERROR, "invalid pixmap depth\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (xoffset) {
+        av_log_ask_for_sample(avctx, "unsupported xoffset %d\n", xoffset);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    if (be > 1) {
+        av_log(avctx, AV_LOG_ERROR, "invalid byte order\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (bitorder > 1) {
+        av_log(avctx, AV_LOG_ERROR, "invalid bitmap bit order\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (bunit != 8 && bunit != 16 && bunit != 32) {
+        av_log(avctx, AV_LOG_ERROR, "invalid bitmap unit\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (bpad != 8 && bpad != 16 && bpad != 32) {
+        av_log(avctx, AV_LOG_ERROR, "invalid bitmap scan-line pad\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (bpp == 0 || bpp > 32) {
+        av_log(avctx, AV_LOG_ERROR, "invalid bits per pixel\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (ncolors > 256) {
+        av_log(avctx, AV_LOG_ERROR, "invalid number of entries in colormap\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if ((ret = av_image_check_size(avctx->width, avctx->height, 0, NULL)) < 0)
+        return ret;
+
+    rsize = FFALIGN(avctx->width * bpp, bpad) / 8;
+    if (lsize < rsize) {
+        av_log(avctx, AV_LOG_ERROR, "invalid bytes per scan-line\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (buf_size < header_size + ncolors * XWD_CMAP_SIZE + avctx->height * lsize) {
+        av_log(avctx, AV_LOG_ERROR, "input buffer too small\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (pixformat != XWD_Z_PIXMAP) {
+        av_log(avctx, AV_LOG_ERROR, "pixmap format %d unsupported\n", pixformat);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avctx->pix_fmt = PIX_FMT_NONE;
+    switch (vclass) {
+    case XWD_STATIC_GRAY:
+    case XWD_GRAY_SCALE:
+        if (bpp != 1)
+            return AVERROR_INVALIDDATA;
+        if (pixdepth == 1)
+            avctx->pix_fmt = PIX_FMT_MONOWHITE;
+        break;
+    case XWD_STATIC_COLOR:
+    case XWD_PSEUDO_COLOR:
+        if (bpp == 8)
+            avctx->pix_fmt = PIX_FMT_PAL8;
+        break;
+    case XWD_TRUE_COLOR:
+    case XWD_DIRECT_COLOR:
+        if (bpp != 16 && bpp != 24 && bpp != 32)
+            return AVERROR_INVALIDDATA;
+        if (bpp == 16 && pixdepth == 15) {
+            if (rgb[0] == 0x7C00 && rgb[1] == 0x3E0 && rgb[2] == 0x1F)
+                avctx->pix_fmt = be ? PIX_FMT_RGB555BE : PIX_FMT_RGB555LE;
+            else if (rgb[0] == 0x1F && rgb[1] == 0x3E0 && rgb[2] == 0x7C00)
+                avctx->pix_fmt = be ? PIX_FMT_BGR555BE : PIX_FMT_BGR555LE;
+        } else if (bpp == 16 && pixdepth == 16) {
+            if (rgb[0] == 0xF800 && rgb[1] == 0x7E0 && rgb[2] == 0x1F)
+                avctx->pix_fmt = be ? PIX_FMT_RGB565BE : PIX_FMT_RGB565LE;
+            else if (rgb[0] == 0x1F && rgb[1] == 0x7E0 && rgb[2] == 0xF800)
+                avctx->pix_fmt = be ? PIX_FMT_BGR565BE : PIX_FMT_BGR565LE;
+        } else if (bpp == 24) {
+            if (rgb[0] == 0xFF0000 && rgb[1] == 0xFF00 && rgb[2] == 0xFF)
+                avctx->pix_fmt = be ? PIX_FMT_RGB24 : PIX_FMT_BGR24;
+            else if (rgb[0] == 0xFF && rgb[1] == 0xFF00 && rgb[2] == 0xFF0000)
+                avctx->pix_fmt = be ? PIX_FMT_BGR24 : PIX_FMT_RGB24;
+        } else if (bpp == 32) {
+            if (rgb[0] == 0xFF0000 && rgb[1] == 0xFF00 && rgb[2] == 0xFF)
+                avctx->pix_fmt = be ? PIX_FMT_ARGB : PIX_FMT_BGRA;
+            else if (rgb[0] == 0xFF && rgb[1] == 0xFF00 && rgb[2] == 0xFF0000)
+                avctx->pix_fmt = be ? PIX_FMT_ABGR : PIX_FMT_RGBA;
+        }
+        buf += ncolors * XWD_CMAP_SIZE;
+        break;
+    default:
+        av_log(avctx, AV_LOG_ERROR, "invalid visual class\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (avctx->pix_fmt == PIX_FMT_NONE) {
+        av_log_ask_for_sample(avctx, "unknown file: bpp %d, pixdepth %d, vclass %d\n", bpp, pixdepth, vclass);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    if (p->data[0])
+        avctx->release_buffer(avctx, p);
+
+    p->reference = 0;
+    if ((ret = avctx->get_buffer(avctx, p)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
+    }
+
+    p->key_frame = 1;
+    p->pict_type = AV_PICTURE_TYPE_I;
+
+    if (avctx->pix_fmt == PIX_FMT_PAL8) {
+        uint32_t *dst = (uint32_t *)p->data[1];
+        uint8_t red, green, blue;
+
+        for (i = 0; i < ncolors; i++) {
+
+            buf   += 4;  // skip colormap entry number
+            red    = *buf; buf += 2;
+            green  = *buf; buf += 2;
+            blue   = *buf; buf += 2;
+            buf   += 2;  // skip bitmask flag and padding
+
+            dst[i] = red << 16 | green << 8 | blue;
+        }
+    }
+
+    ptr = p->data[0];
+    for (i = 0; i < avctx->height; i++) {
+        bytestream_get_buffer(&buf, ptr, rsize);
+        buf += lsize - rsize;
+        ptr += p->linesize[0];
+    }
+
+    *data_size = sizeof(AVFrame);
+    *(AVFrame *)data = *p;
+
+    return buf_size;
+}
+
+static av_cold int xwd_decode_close(AVCodecContext *avctx)
+{
+    if (avctx->coded_frame->data[0])
+        avctx->release_buffer(avctx, avctx->coded_frame);
+
+    av_freep(&avctx->coded_frame);
+
+    return 0;
+}
+
+AVCodec ff_xwd_decoder = {
+    .name           = "xwd",
+    .type           = AVMEDIA_TYPE_VIDEO,
+    .id             = CODEC_ID_XWD,
+    .init           = xwd_decode_init,
+    .close          = xwd_decode_close,
+    .decode         = xwd_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("XWD (X Window Dump) image"),
+};

libavcodec/xwdenc.c

+/*
+ * XWD image format
+ *
+ * Copyright (c) 2012 Paul B Mahol
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "libavutil/pixdesc.h"
+#include "avcodec.h"
+#include "bytestream.h"
+#include "xwd.h"
+
+#define WINDOW_NAME         "lavcxwdenc"
+#define WINDOW_NAME_SIZE    11
+
+static av_cold int xwd_encode_init(AVCodecContext *avctx)
+{
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int xwd_encode_frame(AVCodecContext *avctx, uint8_t *buf,
+                            int buf_size, void *data)
+{
+    AVFrame *p = data;
+    enum PixelFormat pix_fmt = avctx->pix_fmt;
+    uint32_t pixdepth, bpp, bpad, ncolors = 0, lsize, vclass, be = 0;
+    uint32_t rgb[3] = { 0 };
+    uint32_t header_size;
+    int i, out_size;
+    uint8_t *ptr;
+
+    pixdepth = av_get_bits_per_pixel(&av_pix_fmt_descriptors[pix_fmt]);
+    if (av_pix_fmt_descriptors[pix_fmt].flags & PIX_FMT_BE)
+        be = 1;
+    switch (pix_fmt) {
+    case PIX_FMT_ARGB:
+    case PIX_FMT_BGRA:
+    case PIX_FMT_RGBA:
+    case PIX_FMT_ABGR:
+        if (pix_fmt == PIX_FMT_ARGB ||
+            pix_fmt == PIX_FMT_ABGR)
+            be = 1;
+        if (pix_fmt == PIX_FMT_ABGR ||
+            pix_fmt == PIX_FMT_RGBA) {
+            rgb[0] = 0xFF;
+            rgb[1] = 0xFF00;
+            rgb[2] = 0xFF0000;
+        } else {
+            rgb[0] = 0xFF0000;
+            rgb[1] = 0xFF00;
+            rgb[2] = 0xFF;
+        }
+        bpp      = 32;
+        pixdepth = 24;
+        vclass   = XWD_TRUE_COLOR;
+        bpad     = 32;
+        break;
+    case PIX_FMT_BGR24:
+    case PIX_FMT_RGB24:
+        if (pix_fmt == PIX_FMT_RGB24)
+            be = 1;
+        bpp      = 24;
+        vclass   = XWD_TRUE_COLOR;
+        bpad     = 32;
+        rgb[0]   = 0xFF0000;
+        rgb[1]   = 0xFF00;
+        rgb[2]   = 0xFF;
+        break;
+    case PIX_FMT_RGB565LE:
+    case PIX_FMT_RGB565BE:
+    case PIX_FMT_BGR565LE:
+    case PIX_FMT_BGR565BE:
+        if (pix_fmt == PIX_FMT_BGR565LE ||
+            pix_fmt == PIX_FMT_BGR565BE) {
+            rgb[0] = 0x1F;
+            rgb[1] = 0x7E0;
+            rgb[2] = 0xF800;
+        } else {
+            rgb[0] = 0xF800;
+            rgb[1] = 0x7E0;
+            rgb[2] = 0x1F;
+        }
+        bpp      = 16;
+        vclass   = XWD_TRUE_COLOR;
+        bpad     = 16;
+        break;
+    case PIX_FMT_RGB555LE:
+    case PIX_FMT_RGB555BE:
+    case PIX_FMT_BGR555LE:
+    case PIX_FMT_BGR555BE:
+        if (pix_fmt == PIX_FMT_BGR555LE ||
+            pix_fmt == PIX_FMT_BGR555BE) {
+            rgb[0] = 0x1F;
+            rgb[1] = 0x3E0;
+            rgb[2] = 0x7C00;
+        } else {
+            rgb[0] = 0x7C00;
+            rgb[1] = 0x3E0;
+            rgb[2] = 0x1F;
+        }
+        bpp      = 16;
+        vclass   = XWD_TRUE_COLOR;
+        bpad     = 16;
+        break;
+    case PIX_FMT_RGB8:
+    case PIX_FMT_BGR8:
+    case PIX_FMT_RGB4_BYTE:
+    case PIX_FMT_BGR4_BYTE:
+    case PIX_FMT_PAL8:
+        bpp      = 8;
+        vclass   = XWD_PSEUDO_COLOR;
+        bpad     = 8;
+        ncolors  = 256;
+        break;
+    case PIX_FMT_MONOWHITE:
+        bpp      = 1;
+        bpad     = 8;
+        vclass   = XWD_STATIC_GRAY;
+        break;
+    default:
+        av_log(avctx, AV_LOG_INFO, "unsupported pixel format\n");
+        return AVERROR(EINVAL);
+    }
+
+    lsize       = FFALIGN(bpp * avctx->width, bpad) / 8;
+    header_size = XWD_HEADER_SIZE + WINDOW_NAME_SIZE;
+    out_size    = header_size + ncolors * XWD_CMAP_SIZE + avctx->height * lsize;
+
+    if (buf_size < out_size) {
+        av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+        return AVERROR(ENOMEM);
+    }
+
+    avctx->coded_frame->key_frame = 1;
+    avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
+
+    bytestream_put_be32(&buf, header_size);
+    bytestream_put_be32(&buf, XWD_VERSION);   // file version
+    bytestream_put_be32(&buf, XWD_Z_PIXMAP);  // pixmap format
+    bytestream_put_be32(&buf, pixdepth);      // pixmap depth in pixels
+    bytestream_put_be32(&buf, avctx->width);  // pixmap width in pixels
+    bytestream_put_be32(&buf, avctx->height); // pixmap height in pixels
+    bytestream_put_be32(&buf, 0);             // bitmap x offset
+    bytestream_put_be32(&buf, be);            // byte order
+    bytestream_put_be32(&buf, 32);            // bitmap unit
+    bytestream_put_be32(&buf, be);            // bit-order of image data
+    bytestream_put_be32(&buf, bpad);          // bitmap scan-line pad in bits
+    bytestream_put_be32(&buf, bpp);           // bits per pixel
+    bytestream_put_be32(&buf, lsize);         // bytes per scan-line
+    bytestream_put_be32(&buf, vclass);        // visual class
+    bytestream_put_be32(&buf, rgb[0]);        // red mask
+    bytestream_put_be32(&buf, rgb[1]);        // green mask
+    bytestream_put_be32(&buf, rgb[2]);        // blue mask
+    bytestream_put_be32(&buf, 8);             // size of each bitmask in bits
+    bytestream_put_be32(&buf, ncolors);       // number of colors
+    bytestream_put_be32(&buf, ncolors);       // number of entries in color map
+    bytestream_put_be32(&buf, avctx->width);  // window width
+    bytestream_put_be32(&buf, avctx->height); // window height
+    bytestream_put_be32(&buf, 0);             // window upper left X coordinate
+    bytestream_put_be32(&buf, 0);             // window upper left Y coordinate
+    bytestream_put_be32(&buf, 0);             // window border width
+    bytestream_put_buffer(&buf, WINDOW_NAME, WINDOW_NAME_SIZE);
+
+    for (i = 0; i < ncolors; i++) {
+        uint32_t val;
+        uint8_t red, green, blue;
+
+        val   = AV_RN32A(p->data[1] + i * 4);
+        red   = (val >> 16) & 0xFF;
+        green = (val >>  8) & 0xFF;
+        blue  =  val        & 0xFF;
+
+        bytestream_put_be32(&buf, i);         // colormap entry number
+        bytestream_put_be16(&buf, red   << 8);
+        bytestream_put_be16(&buf, green << 8);
+        bytestream_put_be16(&buf, blue  << 8);
+        bytestream_put_byte(&buf, 0x7);       // bitmask flag
+        bytestream_put_byte(&buf, 0);         // padding
+    }
+
+    ptr = p->data[0];
+    for (i = 0; i < avctx->height; i++) {
+        bytestream_put_buffer(&buf, ptr, lsize);
+        ptr += p->linesize[0];
+    }
+
+    return out_size;
+}
+
+static av_cold int xwd_encode_close(AVCodecContext *avctx)
+{
+    av_freep(&avctx->coded_frame);
+
+    return 0;
+}
+
+AVCodec ff_xwd_encoder = {
+    .name         = "xwd",
+    .type         = AVMEDIA_TYPE_VIDEO,
+    .id           = CODEC_ID_XWD,
+    .init         = xwd_encode_init,
+    .encode       = xwd_encode_frame,
+    .close        = xwd_encode_close,
+    .pix_fmts     = (const enum PixelFormat[]) { PIX_FMT_BGRA,
+                                                 PIX_FMT_RGBA,
+                                                 PIX_FMT_ARGB,
+                                                 PIX_FMT_ABGR,
+                                                 PIX_FMT_RGB24,
+                                                 PIX_FMT_BGR24,
+                                                 PIX_FMT_RGB565BE,
+                                                 PIX_FMT_RGB565LE,
+                                                 PIX_FMT_BGR565BE,
+                                                 PIX_FMT_BGR565LE,
+                                                 PIX_FMT_RGB555BE,
+                                                 PIX_FMT_RGB555LE,
+                                                 PIX_FMT_BGR555BE,
+                                                 PIX_FMT_BGR555LE,
+                                                 PIX_FMT_RGB8,
+                                                 PIX_FMT_BGR8,
+                                                 PIX_FMT_RGB4_BYTE,
+                                                 PIX_FMT_BGR4_BYTE,
+                                                 PIX_FMT_PAL8,
+                                                 PIX_FMT_MONOWHITE,
+                                                 PIX_FMT_NONE },
+    .long_name    = NULL_IF_CONFIG_SMALL("XWD (X Window Dump) image"),
+};

libavformat/Makefile

@@ -326,8 +326,8 @@ OBJS-$(CONFIG_WC3_DEMUXER)               += wc3movie.o
 OBJS-$(CONFIG_WEBM_MUXER)                += matroskaenc.o matroska.o \
                                             riff.o isom.o avc.o \
                                             flacenc_header.o avlanguage.o
-OBJS-$(CONFIG_WSAUD_DEMUXER)             += westwood.o
-OBJS-$(CONFIG_WSVQA_DEMUXER)             += westwood.o
+OBJS-$(CONFIG_WSAUD_DEMUXER)             += westwood_aud.o
+OBJS-$(CONFIG_WSVQA_DEMUXER)             += westwood_vqa.o
 OBJS-$(CONFIG_WTV_DEMUXER)               += wtvdec.o wtv.o asfdec.o asf.o asfcrypt.o \
                                             avlanguage.o mpegts.o isom.o riff.o
 OBJS-$(CONFIG_WTV_MUXER)                 += wtvenc.o wtv.o asf.o asfenc.o riff.o

libavformat/img2.c

@@ -91,6 +91,7 @@ static const IdStrMap img_tags[] = {
     { CODEC_ID_JPEG2000  , "jpc"},
     { CODEC_ID_DPX       , "dpx"},
     { CODEC_ID_PICTOR    , "pic"},
+    { CODEC_ID_XWD       , "xwd"},
     { CODEC_ID_NONE      , NULL}
 };
 
@@ -528,7 +529,7 @@ AVOutputFormat ff_image2_muxer = {
     .name           = "image2",
     .long_name      = NULL_IF_CONFIG_SMALL("image2 sequence"),
     .extensions     = "bmp,dpx,jls,jpeg,jpg,ljpg,pam,pbm,pcx,pgm,pgmyuv,png,"
-                      "ppm,sgi,tga,tif,tiff,jp2,j2c",
+                      "ppm,sgi,tga,tif,tiff,jp2,j2c,xwd",
     .priv_data_size = sizeof(VideoData),
     .video_codec    = CODEC_ID_MJPEG,
     .write_header   = write_header,

libavformat/westwood_aud.c

+/*
+ * Westwood Studios AUD Format Demuxer
+ * Copyright (c) 2003 The ffmpeg Project
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Westwood Studios AUD file demuxer
+ * by Mike Melanson (melanson@pcisys.net)
+ * for more information on the Westwood file formats, visit:
+ *   http://www.pcisys.net/~melanson/codecs/
+ *   http://www.geocities.com/SiliconValley/8682/aud3.txt
+ *
+ * Implementation note: There is no definite file signature for AUD files.
+ * The demuxer uses a probabilistic strategy for content detection. This
+ * entails performing sanity checks on certain header values in order to
+ * qualify a file. Refer to wsaud_probe() for the precise parameters.
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "internal.h"
+
+#define AUD_HEADER_SIZE 12
+#define AUD_CHUNK_PREAMBLE_SIZE 8
+#define AUD_CHUNK_SIGNATURE 0x0000DEAF
+
+typedef struct WsAudDemuxContext {
+    int audio_samplerate;
+    int audio_channels;
+    int audio_bits;
+    enum CodecID audio_type;
+    int audio_stream_index;
+    int64_t audio_frame_counter;
+} WsAudDemuxContext;
+
+static int wsaud_probe(AVProbeData *p)
+{
+    int field;
+
+    /* Probabilistic content detection strategy: There is no file signature
+     * so perform sanity checks on various header parameters:
+     *   8000 <= sample rate (16 bits) <= 48000  ==> 40001 acceptable numbers
+     *   flags <= 0x03 (2 LSBs are used)         ==> 4 acceptable numbers
+     *   compression type (8 bits) = 1 or 99     ==> 2 acceptable numbers
+     *   first audio chunk signature (32 bits)   ==> 1 acceptable number
+     * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
+     * 320008 acceptable number combinations.
+     */
+
+    if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
+        return 0;
+
+    /* check sample rate */
+    field = AV_RL16(&p->buf[0]);
+    if ((field < 8000) || (field > 48000))
+        return 0;
+
+    /* enforce the rule that the top 6 bits of this flags field are reserved (0);
+     * this might not be true, but enforce it until deemed unnecessary */
+    if (p->buf[10] & 0xFC)
+        return 0;
+
+    /* note: only check for WS IMA (type 99) right now since there is no
+     * support for type 1 */
+    if (p->buf[11] != 99)
+        return 0;
+
+    /* read ahead to the first audio chunk and validate the first header signature */
+    if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
+        return 0;
+
+    /* return 1/2 certainty since this file check is a little sketchy */
+    return AVPROBE_SCORE_MAX / 2;
+}
+
+static int wsaud_read_header(AVFormatContext *s,
+                             AVFormatParameters *ap)
+{
+    WsAudDemuxContext *wsaud = s->priv_data;
+    AVIOContext *pb = s->pb;
+    AVStream *st;
+    unsigned char header[AUD_HEADER_SIZE];
+
+    if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
+        return AVERROR(EIO);
+    wsaud->audio_samplerate = AV_RL16(&header[0]);
+    if (header[11] == 99)
+        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
+    else
+        return AVERROR_INVALIDDATA;
+
+    /* flag 0 indicates stereo */
+    wsaud->audio_channels = (header[10] & 0x1) + 1;
+    /* flag 1 indicates 16 bit audio */
+    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
+
+    /* initialize the audio decoder stream */
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+    avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
+    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+    st->codec->codec_id = wsaud->audio_type;
+    st->codec->codec_tag = 0;  /* no tag */
+    st->codec->channels = wsaud->audio_channels;
+    st->codec->sample_rate = wsaud->audio_samplerate;
+    st->codec->bits_per_coded_sample = wsaud->audio_bits;
+    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
+        st->codec->bits_per_coded_sample / 4;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+
+    wsaud->audio_stream_index = st->index;
+    wsaud->audio_frame_counter = 0;
+
+    return 0;
+}
+
+static int wsaud_read_packet(AVFormatContext *s,
+                             AVPacket *pkt)
+{
+    WsAudDemuxContext *wsaud = s->priv_data;
+    AVIOContext *pb = s->pb;
+    unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
+    unsigned int chunk_size;
+    int ret = 0;
+
+    if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
+        AUD_CHUNK_PREAMBLE_SIZE)
+        return AVERROR(EIO);
+
+    /* validate the chunk */
+    if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
+        return AVERROR_INVALIDDATA;
+
+    chunk_size = AV_RL16(&preamble[0]);
+    ret= av_get_packet(pb, pkt, chunk_size);
+    if (ret != chunk_size)
+        return AVERROR(EIO);
+    pkt->stream_index = wsaud->audio_stream_index;
+    pkt->pts = wsaud->audio_frame_counter;
+    pkt->pts /= wsaud->audio_samplerate;
+
+    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
+    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
+
+    return ret;
+}
+
+AVInputFormat ff_wsaud_demuxer = {
+    .name           = "wsaud",
+    .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
+    .priv_data_size = sizeof(WsAudDemuxContext),
+    .read_probe     = wsaud_probe,
+    .read_header    = wsaud_read_header,
+    .read_packet    = wsaud_read_packet,
+};

libavformat/westwood.c → libavformat/westwood_vqa.c

 /*
- * Westwood Studios Multimedia Formats Demuxer (VQA, AUD)
+ * Westwood Studios VQA Format Demuxer
  * Copyright (c) 2003 The ffmpeg Project
  *
  * This file is part of FFmpeg.
@@ -21,26 +21,17 @@
 
 /**
  * @file
- * Westwood Studios VQA & AUD file demuxers
+ * Westwood Studios VQA file demuxer
  * by Mike Melanson (melanson@pcisys.net)
  * for more information on the Westwood file formats, visit:
  *   http://www.pcisys.net/~melanson/codecs/
  *   http://www.geocities.com/SiliconValley/8682/aud3.txt
- *
- * Implementation note: There is no definite file signature for AUD files.
- * The demuxer uses a probabilistic strategy for content detection. This
- * entails performing sanity checks on certain header values in order to
- * qualify a file. Refer to wsaud_probe() for the precise parameters.
  */
 
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
 
-#define AUD_HEADER_SIZE 12
-#define AUD_CHUNK_PREAMBLE_SIZE 8
-#define AUD_CHUNK_SIGNATURE 0x0000DEAF
-
 #define FORM_TAG MKBETAG('F', 'O', 'R', 'M')
 #define WVQA_TAG MKBETAG('W', 'V', 'Q', 'A')
 #define VQHD_TAG MKBETAG('V', 'Q', 'H', 'D')
@@ -63,15 +54,6 @@
 #define VQA_FRAMERATE 15
 #define VQA_PREAMBLE_SIZE 8
 
-typedef struct WsAudDemuxContext {
-    int audio_samplerate;
-    int audio_channels;
-    int audio_bits;
-    enum CodecID audio_type;
-    int audio_stream_index;
-    int64_t audio_frame_counter;
-} WsAudDemuxContext;
-
 typedef struct WsVqaDemuxContext {
     int audio_samplerate;
     int audio_channels;
@@ -83,119 +65,6 @@ typedef struct WsVqaDemuxContext {
     int64_t audio_frame_counter;
 } WsVqaDemuxContext;
 
-static int wsaud_probe(AVProbeData *p)
-{
-    int field;
-
-    /* Probabilistic content detection strategy: There is no file signature
-     * so perform sanity checks on various header parameters:
-     *   8000 <= sample rate (16 bits) <= 48000  ==> 40001 acceptable numbers
-     *   flags <= 0x03 (2 LSBs are used)         ==> 4 acceptable numbers
-     *   compression type (8 bits) = 1 or 99     ==> 2 acceptable numbers
-     *   first audio chunk signature (32 bits)   ==> 1 acceptable number
-     * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
-     * 320008 acceptable number combinations.
-     */
-
-    if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
-        return 0;
-
-    /* check sample rate */
-    field = AV_RL16(&p->buf[0]);
-    if ((field < 8000) || (field > 48000))
-        return 0;
-
-    /* enforce the rule that the top 6 bits of this flags field are reserved (0);
-     * this might not be true, but enforce it until deemed unnecessary */
-    if (p->buf[10] & 0xFC)
-        return 0;
-
-    /* note: only check for WS IMA (type 99) right now since there is no
-     * support for type 1 */
-    if (p->buf[11] != 99)
-        return 0;
-
-    /* read ahead to the first audio chunk and validate the first header signature */
-    if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
-        return 0;
-
-    /* return 1/2 certainty since this file check is a little sketchy */
-    return AVPROBE_SCORE_MAX / 2;
-}
-
-static int wsaud_read_header(AVFormatContext *s,
-                             AVFormatParameters *ap)
-{
-    WsAudDemuxContext *wsaud = s->priv_data;
-    AVIOContext *pb = s->pb;
-    AVStream *st;
-    unsigned char header[AUD_HEADER_SIZE];
-
-    if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
-        return AVERROR(EIO);
-    wsaud->audio_samplerate = AV_RL16(&header[0]);
-    if (header[11] == 99)
-        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
-    else
-        return AVERROR_INVALIDDATA;
-
-    /* flag 0 indicates stereo */
-    wsaud->audio_channels = (header[10] & 0x1) + 1;
-    /* flag 1 indicates 16 bit audio */
-    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
-
-    /* initialize the audio decoder stream */
-    st = avformat_new_stream(s, NULL);
-    if (!st)
-        return AVERROR(ENOMEM);
-    avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
-    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = wsaud->audio_type;
-    st->codec->codec_tag = 0;  /* no tag */
-    st->codec->channels = wsaud->audio_channels;
-    st->codec->sample_rate = wsaud->audio_samplerate;
-    st->codec->bits_per_coded_sample = wsaud->audio_bits;
-    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-        st->codec->bits_per_coded_sample / 4;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
-
-    wsaud->audio_stream_index = st->index;
-    wsaud->audio_frame_counter = 0;
-
-    return 0;
-}
-
-static int wsaud_read_packet(AVFormatContext *s,
-                             AVPacket *pkt)
-{
-    WsAudDemuxContext *wsaud = s->priv_data;
-    AVIOContext *pb = s->pb;
-    unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
-    unsigned int chunk_size;
-    int ret = 0;
-
-    if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
-        AUD_CHUNK_PREAMBLE_SIZE)
-        return AVERROR(EIO);
-
-    /* validate the chunk */
-    if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
-        return AVERROR_INVALIDDATA;
-
-    chunk_size = AV_RL16(&preamble[0]);
-    ret= av_get_packet(pb, pkt, chunk_size);
-    if (ret != chunk_size)
-        return AVERROR(EIO);
-    pkt->stream_index = wsaud->audio_stream_index;
-    pkt->pts = wsaud->audio_frame_counter;
-    pkt->pts /= wsaud->audio_samplerate;
-
-    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
-    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
-
-    return ret;
-}
-
 static int wsvqa_probe(AVProbeData *p)
 {
     /* need 12 bytes to qualify */
@@ -367,17 +236,6 @@ static int wsvqa_read_packet(AVFormatContext *s,
     return ret;
 }
 
-#if CONFIG_WSAUD_DEMUXER
-AVInputFormat ff_wsaud_demuxer = {
-    .name           = "wsaud",
-    .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
-    .priv_data_size = sizeof(WsAudDemuxContext),
-    .read_probe     = wsaud_probe,
-    .read_header    = wsaud_read_header,
-    .read_packet    = wsaud_read_packet,
-};
-#endif
-#if CONFIG_WSVQA_DEMUXER
 AVInputFormat ff_wsvqa_demuxer = {
     .name           = "wsvqa",
     .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"),
@@ -386,4 +244,3 @@ AVInputFormat ff_wsvqa_demuxer = {
     .read_header    = wsvqa_read_header,
     .read_packet    = wsvqa_read_packet,
 };
-#endif