Merge remote-tracking branch 'qatar/master'

* qatar/master:
  wmaenc: use float planar sample format
  (e)ac3enc: use planar sample format
  aacenc: use planar sample format
  adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
  adpcmenc: move 'ch' variable to higher scope
  adpcmenc: fix 3 instances of variable shadowing
  adpcm_ima_wav: simplify encoding
  libvorbis: use planar sample format
  libmp3lame: use planar sample formats
  vorbisenc: use float planar sample format
  ffm: do not write or read the audio sample format
  parseutils: fix parsing of invalid alpha values
  doc/RELEASE_NOTES: update for the 9 release.
  smoothstreamingenc: Add a more verbose error message
  smoothstreamingenc: Ignore the return value from mkdir
  smoothstreamingenc: Try writing a manifest when opening the muxer
  smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
  smoothstreamingenc: Properly return errors from ism_flush to the caller
  smoothstreamingenc: Check the output UrlContext before accessing it

Conflicts:
	doc/RELEASE_NOTES
	libavcodec/aacenc.c
	libavcodec/ac3enc_template.c
	libavcodec/wmaenc.c
	tests/ref/lavf/ffm

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

1 files changed, 72 insertions, 81 deletions

diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c
index a3194abb3c..f016ebd910 100644
--- a/libavcodec/adpcmenc.c
+++ b/libavcodec/adpcmenc.c
@@ -274,12 +274,11 @@ static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
 
 static void adpcm_compress_trellis(AVCodecContext *avctx,
                                    const int16_t *samples, uint8_t *dst,
-                                   ADPCMChannelStatus *c, int n)
+                                   ADPCMChannelStatus *c, int n, int stride)
 {
     //FIXME 6% faster if frontier is a compile-time constant
     ADPCMEncodeContext *s = avctx->priv_data;
     const int frontier = 1 << avctx->trellis;
-    const int stride   = avctx->channels;
     const int version  = avctx->codec->id;
     TrellisPath *paths       = s->paths, *p;
     TrellisNode *node_buf    = s->node_buf;
@@ -481,13 +480,15 @@ static void adpcm_compress_trellis(AVCodecContext *avctx,
 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                               const AVFrame *frame, int *got_packet_ptr)
 {
-    int n, i, st, pkt_size, ret;
+    int n, i, ch, st, pkt_size, ret;
     const int16_t *samples;
+    int16_t **samples_p;
     uint8_t *dst;
     ADPCMEncodeContext *c = avctx->priv_data;
     uint8_t *buf;
 
     samples = (const int16_t *)frame->data[0];
+    samples_p = (int16_t **)frame->extended_data;
     st = avctx->channels == 2;
 
     if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
@@ -500,91 +501,71 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
     switch(avctx->codec->id) {
     case AV_CODEC_ID_ADPCM_IMA_WAV:
-        n = frame->nb_samples / 8;
-        c->status[0].prev_sample = samples[0];
-        /* c->status[0].step_index = 0;
-        XXX: not sure how to init the state machine */
-        bytestream_put_le16(&dst, c->status[0].prev_sample);
-        *dst++ = c->status[0].step_index;
-        *dst++ = 0; /* unknown */
-        samples++;
-        if (avctx->channels == 2) {
-            c->status[1].prev_sample = samples[0];
-            /* c->status[1].step_index = 0; */
-            bytestream_put_le16(&dst, c->status[1].prev_sample);
-            *dst++ = c->status[1].step_index;
-            *dst++ = 0;
-            samples++;
+    {
+        int blocks, j;
+
+        blocks = (frame->nb_samples - 1) / 8;
+
+        for (ch = 0; ch < avctx->channels; ch++) {
+            ADPCMChannelStatus *status = &c->status[ch];
+            status->prev_sample = samples_p[ch][0];
+            /* status->step_index = 0;
+               XXX: not sure how to init the state machine */
+            bytestream_put_le16(&dst, status->prev_sample);
+            *dst++ = status->step_index;
+            *dst++ = 0; /* unknown */
         }
 
-        /* stereo: 4 bytes (8 samples) for left,
-            4 bytes for right, 4 bytes left, ... */
+        /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
         if (avctx->trellis > 0) {
-            FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 8, error);
-            adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n * 8);
-            if (avctx->channels == 2)
-                adpcm_compress_trellis(avctx, samples + 1, buf + n * 8,
-                                       &c->status[1], n * 8);
-            for (i = 0; i < n; i++) {
-                *dst++ = buf[8 * i + 0] | (buf[8 * i + 1] << 4);
-                *dst++ = buf[8 * i + 2] | (buf[8 * i + 3] << 4);
-                *dst++ = buf[8 * i + 4] | (buf[8 * i + 5] << 4);
-                *dst++ = buf[8 * i + 6] | (buf[8 * i + 7] << 4);
-                if (avctx->channels == 2) {
-                    uint8_t *buf1 = buf + n * 8;
-                    *dst++ = buf1[8 * i + 0] | (buf1[8 * i + 1] << 4);
-                    *dst++ = buf1[8 * i + 2] | (buf1[8 * i + 3] << 4);
-                    *dst++ = buf1[8 * i + 4] | (buf1[8 * i + 5] << 4);
-                    *dst++ = buf1[8 * i + 6] | (buf1[8 * i + 7] << 4);
+            FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
+            for (ch = 0; ch < avctx->channels; ch++) {
+                adpcm_compress_trellis(avctx, &samples_p[ch][1],
+                                       buf + ch * blocks * 8, &c->status[ch],
+                                       blocks * 8, 1);
+            }
+            for (i = 0; i < blocks; i++) {
+                for (ch = 0; ch < avctx->channels; ch++) {
+                    uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
+                    for (j = 0; j < 8; j += 2)
+                        *dst++ = buf1[j] | (buf1[j + 1] << 4);
                 }
             }
             av_free(buf);
         } else {
-            for (; n > 0; n--) {
-                *dst    = adpcm_ima_compress_sample(&c->status[0], samples[0]);
-                *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels    ]) << 4;
-                *dst    = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
-                *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
-                *dst    = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
-                *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
-                *dst    = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
-                *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
-                /* right channel */
-                if (avctx->channels == 2) {
-                    *dst    = adpcm_ima_compress_sample(&c->status[1], samples[1 ]);
-                    *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[3 ]) << 4;
-                    *dst    = adpcm_ima_compress_sample(&c->status[1], samples[5 ]);
-                    *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[7 ]) << 4;
-                    *dst    = adpcm_ima_compress_sample(&c->status[1], samples[9 ]);
-                    *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
-                    *dst    = adpcm_ima_compress_sample(&c->status[1], samples[13]);
-                    *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
+            for (i = 0; i < blocks; i++) {
+                for (ch = 0; ch < avctx->channels; ch++) {
+                    ADPCMChannelStatus *status = &c->status[ch];
+                    const int16_t *smp = &samples_p[ch][1 + i * 8];
+                    for (j = 0; j < 8; j += 2) {
+                        *dst++ = adpcm_ima_compress_sample(status, smp[j    ]) |
+                                (adpcm_ima_compress_sample(status, smp[j + 1]) << 4);
+                    }
                 }
-                samples += 8 * avctx->channels;
             }
         }
         break;
+    }
     case AV_CODEC_ID_ADPCM_IMA_QT:
     {
-        int ch, i;
         PutBitContext pb;
         init_put_bits(&pb, dst, pkt_size * 8);
 
         for (ch = 0; ch < avctx->channels; ch++) {
-            put_bits(&pb, 9, (c->status[ch].prev_sample & 0xFFFF) >> 7);
-            put_bits(&pb, 7,  c->status[ch].step_index);
+            ADPCMChannelStatus *status = &c->status[ch];
+            put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
+            put_bits(&pb, 7,  status->step_index);
             if (avctx->trellis > 0) {
                 uint8_t buf[64];
-                adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
+                adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
+                                       64, 1);
                 for (i = 0; i < 64; i++)
                     put_bits(&pb, 4, buf[i ^ 1]);
             } else {
                 for (i = 0; i < 64; i += 2) {
                     int t1, t2;
-                    t1 = adpcm_ima_qt_compress_sample(&c->status[ch],
-                                                      samples[avctx->channels * (i + 0) + ch]);
-                    t2 = adpcm_ima_qt_compress_sample(&c->status[ch],
-                                                      samples[avctx->channels * (i + 1) + ch]);
+                    t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i    ]);
+                    t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
                     put_bits(&pb, 4, t2);
                     put_bits(&pb, 4, t1);
                 }
@@ -596,7 +577,6 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     }
     case AV_CODEC_ID_ADPCM_SWF:
     {
-        int i;
         PutBitContext pb;
         init_put_bits(&pb, dst, pkt_size * 8);
 
@@ -617,10 +597,11 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         if (avctx->trellis > 0) {
             FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
             adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
-                                   &c->status[0], n);
+                                   &c->status[0], n, avctx->channels);
             if (avctx->channels == 2)
                 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
-                                       buf + n, &c->status[1], n);
+                                       buf + n, &c->status[1], n,
+                                       avctx->channels);
             for (i = 0; i < n; i++) {
                 put_bits(&pb, 4, buf[i]);
                 if (avctx->channels == 2)
@@ -661,15 +642,18 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             bytestream_put_le16(&dst, c->status[i].sample2);
 
         if (avctx->trellis > 0) {
-            int n = avctx->block_align - 7 * avctx->channels;
+            n = avctx->block_align - 7 * avctx->channels;
             FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
             if (avctx->channels == 1) {
-                adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+                adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
+                                       avctx->channels);
                 for (i = 0; i < n; i += 2)
                     *dst++ = (buf[i] << 4) | buf[i + 1];
             } else {
-                adpcm_compress_trellis(avctx, samples,     buf,     &c->status[0], n);
-                adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
+                adpcm_compress_trellis(avctx, samples,     buf,
+                                       &c->status[0], n, avctx->channels);
+                adpcm_compress_trellis(avctx, samples + 1, buf + n,
+                                       &c->status[1], n, avctx->channels);
                 for (i = 0; i < n; i++)
                     *dst++ = (buf[i] << 4) | buf[n + i];
             }
@@ -689,12 +673,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
             n *= 2;
             if (avctx->channels == 1) {
-                adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+                adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
+                                       avctx->channels);
                 for (i = 0; i < n; i += 2)
                     *dst++ = buf[i] | (buf[i + 1] << 4);
             } else {
-                adpcm_compress_trellis(avctx, samples,     buf,     &c->status[0], n);
-                adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
+                adpcm_compress_trellis(avctx, samples,     buf,
+                                       &c->status[0], n, avctx->channels);
+                adpcm_compress_trellis(avctx, samples + 1, buf + n,
+                                       &c->status[1], n, avctx->channels);
                 for (i = 0; i < n; i++)
                     *dst++ = buf[i] | (buf[n + i] << 4);
             }
@@ -722,7 +709,11 @@ static const enum AVSampleFormat sample_fmts[] = {
     AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
 };
 
-#define ADPCM_ENCODER(id_, name_, long_name_)               \
+static const enum AVSampleFormat sample_fmts_p[] = {
+    AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
+};
+
+#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
 AVCodec ff_ ## name_ ## _encoder = {                        \
     .name           = #name_,                               \
     .type           = AVMEDIA_TYPE_AUDIO,                   \
@@ -731,12 +722,12 @@ AVCodec ff_ ## name_ ## _encoder = {                        \
     .init           = adpcm_encode_init,                    \
     .encode2        = adpcm_encode_frame,                   \
     .close          = adpcm_encode_close,                   \
-    .sample_fmts    = sample_fmts,                          \
+    .sample_fmts    = sample_fmts_,                         \
     .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
 }
 
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt,   "ADPCM IMA QuickTime");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms,           "ADPCM Microsoft");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf,         "ADPCM Shockwave Flash");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha,   "ADPCM Yamaha");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT,  adpcm_ima_qt,  sample_fmts_p, "ADPCM IMA QuickTime");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS,      adpcm_ms,      sample_fmts,   "ADPCM Microsoft");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF,     adpcm_swf,     sample_fmts,   "ADPCM Shockwave Flash");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA,  adpcm_yamaha,  sample_fmts,   "ADPCM Yamaha");