ad_ffmpeg: add support for planar sample formats

FFmpeg and Libav are starting to return a growing number of planar samples
when decoding formats that save data like that.

In this first implementation planar formats are immediately converted to
packed formats. Fututre developments should move to use libavresample.

Original work by Nicolas George on mplayer(1).

1 files changed, 36 insertions, 7 deletions

diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index c4d7c13941..cca7725490 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -52,6 +52,7 @@ struct priv {
     AVCodecContext *avctx;
     AVFrame *avframe;
     char *output;
+    char *output_packed; // used by deplanarize to store packed audio samples
     int output_left;
     int unitsize;
     int previous_data_left;  // input demuxer packet data
@@ -71,7 +72,7 @@ static int setup_format(sh_audio_t *sh_audio,
                         const AVCodecContext *lavc_context)
 {
     int sample_format = sh_audio->sample_format;
-    switch (lavc_context->sample_fmt) {
+    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
     case AV_SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
     case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
     case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
@@ -218,7 +219,7 @@ static int init(sh_audio_t *sh_audio)
     if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
         sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
 
-    switch (lavc_context->sample_fmt) {
+    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
     case AV_SAMPLE_FMT_U8:
     case AV_SAMPLE_FMT_S16:
     case AV_SAMPLE_FMT_S32:
@@ -264,6 +265,34 @@ static int control(sh_audio_t *sh, int cmd, void *arg, ...)
     return CONTROL_UNKNOWN;
 }
 
+static av_always_inline void deplanarize(struct sh_audio *sh)
+{
+    struct priv *priv = sh->context;
+
+    size_t bps        = av_get_bytes_per_sample(priv->avctx->sample_fmt);
+    size_t nb_samples = priv->avframe->nb_samples;
+    size_t channels   = priv->avctx->channels;
+    size_t size       = bps * nb_samples * channels;
+
+    if (talloc_get_size(priv->output_packed) != size)
+        priv->output_packed =
+            talloc_realloc_size(priv, priv->output_packed, size);
+
+    size_t offset = 0;
+    unsigned char *output_ptr = priv->output_packed;
+    unsigned char **src = priv->avframe->data;
+
+    for (size_t s = 0; s < nb_samples; s++) {
+        for (size_t c = 0; c < channels; c++) {
+            memcpy(output_ptr, src[c] + offset, bps);
+            output_ptr += bps;
+        }
+        offset += bps;
+    }
+
+    priv->output = priv->output_packed;
+}
+
 static int decode_new_packet(struct sh_audio *sh)
 {
     struct priv *priv = sh->context;
@@ -320,10 +349,6 @@ static int decode_new_packet(struct sh_audio *sh)
         priv->previous_data_left = insize - ret;
     if (!got_frame)
         return 0;
-    /* An error is reported later from output format checking, but make
-     * sure we don't crash by overreading first plane. */
-    if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1)
-        return 0;
     uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
                         avctx->channels;
     if (unitsize > 100000)
@@ -333,7 +358,11 @@ static int decode_new_packet(struct sh_audio *sh)
     if (output_left > 500000000)
         abort();
     priv->output_left = output_left;
-    priv->output = priv->avframe->data[0];
+    if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
+        deplanarize(sh);
+    } else {
+        priv->output = priv->avframe->data[0];
+    }
     mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d  \n", insize,
            priv->output_left);
     return 0;