Merge commit 'f5fa03660db16f9d78abc5a626438b4d0b54f563'

* commit 'f5fa03660db16f9d78abc5a626438b4d0b54f563':
  vble: Do not abort decoding when version is not 1
  lavr: do not pass consumed samples as a parameter to ff_audio_resample()
  lavr: correct the documentation for the ff_audio_resample() return value
  lavr: do not pass sample count as a parameter to ff_audio_convert()
  x86: h264_weight: port to cpuflags
  configure: Enable avconv filter dependencies automatically

Conflicts:
	configure
	libavcodec/x86/h264_weight.asm

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

5 files changed, 18 insertions, 19 deletions

diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c
index e9835c8e8b..dcf8a39b06 100644
--- a/libavresample/audio_convert.c
+++ b/libavresample/audio_convert.c
@@ -284,9 +284,10 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
     return ac;
 }
 
-int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len)
+int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
 {
     int use_generic = 1;
+    int len         = in->nb_samples;
 
     /* determine whether to use the optimized function based on pointer and
        samples alignment in both the input and output */
diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h
index 2b8bface7d..bc27223140 100644
--- a/libavresample/audio_convert.h
+++ b/libavresample/audio_convert.h
@@ -72,13 +72,16 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
  * examined to determine whether to use the generic or optimized conversion
  * function (when available).
  *
+ * The number of samples to convert is determined by in->nb_samples. The output
+ * buffer must be large enough to handle this many samples. out->nb_samples is
+ * set by this function before a successful return.
+ *
  * @param ac     AudioConvert context
  * @param out    output audio data
  * @param in     input audio data
- * @param len    number of samples to convert
  * @return       0 on success, negative AVERROR code on failure
  */
-int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len);
+int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
 
 /* arch-specific initialization functions */
 
diff --git a/libavresample/resample.c b/libavresample/resample.c
index f0af1ffd58..381d673717 100644
--- a/libavresample/resample.c
+++ b/libavresample/resample.c
@@ -394,10 +394,9 @@ static int resample(ResampleContext *c, void *dst, const void *src,
     return dst_index;
 }
 
-int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
-                      int *consumed)
+int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
 {
-    int ch, in_samples, in_leftover, out_samples = 0;
+    int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
     int ret = AVERROR(EINVAL);
 
     in_samples  = src ? src->nb_samples : 0;
@@ -430,7 +429,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
     /* resample each channel plane */
     for (ch = 0; ch < c->buffer->channels; ch++) {
         out_samples = resample(c, (void *)dst->data[ch],
-                               (const void *)c->buffer->data[ch], consumed,
+                               (const void *)c->buffer->data[ch], &consumed,
                                c->buffer->nb_samples, dst->allocated_samples,
                                ch + 1 == c->buffer->channels);
     }
@@ -440,7 +439,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
     }
 
     /* drain consumed samples from the internal buffer */
-    ff_audio_data_drain(c->buffer, *consumed);
+    ff_audio_data_drain(c->buffer, consumed);
 
     av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
             in_samples, in_leftover, out_samples, c->buffer->nb_samples);
diff --git a/libavresample/resample.h b/libavresample/resample.h
index b42fdbbaac..7534e26ad4 100644
--- a/libavresample/resample.h
+++ b/libavresample/resample.h
@@ -61,10 +61,8 @@ void ff_audio_resample_free(ResampleContext **c);
  * @param c         ResampleContext
  * @param dst       destination audio data
  * @param src       source audio data
- * @param consumed  number of samples consumed from the source
- * @return          number of samples written to the destination
+ * @return          0 on success, negative AVERROR code on failure
  */
-int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
-                      int *consumed);
+int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src);
 
 #endif /* AVRESAMPLE_RESAMPLE_H */
diff --git a/libavresample/utils.c b/libavresample/utils.c
index ad49e880af..5591f1575e 100644
--- a/libavresample/utils.c
+++ b/libavresample/utils.c
@@ -313,8 +313,8 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
                 if (ret < 0)
                     return ret;
                 av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
-                ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
-                                       current_buffer->nb_samples);
+                ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
+                                       current_buffer);
                 if (ret < 0)
                     return ret;
             } else {
@@ -342,7 +342,6 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
 
     if (avr->resample_needed) {
         AudioData *resample_out;
-        int consumed = 0;
 
         if (!avr->out_convert_needed && direct_output && out_samples > 0)
             resample_out = &output_buffer;
@@ -351,7 +350,7 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
         av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
                 resample_out->name);
         ret = ff_audio_resample(avr->resample, resample_out,
-                                current_buffer, &consumed);
+                                current_buffer);
         if (ret < 0)
             return ret;
 
@@ -381,8 +380,7 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
         if (direct_output && out_samples >= current_buffer->nb_samples) {
             /* convert directly to output */
             av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
-            ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
-                                   current_buffer->nb_samples);
+            ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
             if (ret < 0)
                 return ret;
 
@@ -395,7 +393,7 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
                 return ret;
             av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
             ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
-                                   current_buffer, current_buffer->nb_samples);
+                                   current_buffer);
             if (ret < 0)
                 return ret;
             current_buffer = avr->out_buffer;