decavcodec: prefer AVFrame fields over their AVCodecContext equivalent.

These fields correspond to the actual audio data in the AVFrame, and should always be set (otherwise, we must submit a bug report to libav).


git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5994 b64f7644-9d1e-0410-96f1-a4d463321fa5

10 files changed, 31 insertions, 80 deletions

diff --git a/libhb/audio_remap.c b/libhb/audio_remap.c
index 44f0d73d3..d8c7f3de7 100644
--- a/libhb/audio_remap.c
+++ b/libhb/audio_remap.c
@@ -246,19 +246,25 @@ fail:
 }
 
 void hb_audio_remap_set_channel_layout(hb_audio_remap_t *remap,
-                                       uint64_t channel_layout,
-                                       int channels)
+                                       uint64_t channel_layout)
 {
     if (remap != NULL)
     {
         int ii;
         remap->remap_needed = 0;
 
+        // sanitize the layout
+        if (channel_layout == AV_CH_LAYOUT_STEREO_DOWNMIX)
+        {
+            channel_layout = AV_CH_LAYOUT_STEREO;
+        }
+        remap->nchannels = av_get_channel_layout_nb_channels(channel_layout);
+
         // in some cases, remapping is not necessary and/or supported
-        if (channels > HB_AUDIO_REMAP_MAX_CHANNELS)
+        if (remap->nchannels > HB_AUDIO_REMAP_MAX_CHANNELS)
         {
             hb_log("hb_audio_remap_set_channel_layout: too many channels (%d)",
-                   channels);
+                   remap->nchannels);
             return;
         }
         if (remap->channel_map_in == remap->channel_map_out)
@@ -266,14 +272,6 @@ void hb_audio_remap_set_channel_layout(hb_audio_remap_t *remap,
             return;
         }
 
-        // sanitize the layout
-        channel_layout = hb_ff_layout_xlat(channel_layout, channels);
-        if (channel_layout == AV_CH_LAYOUT_STEREO_DOWNMIX)
-        {
-            channel_layout = AV_CH_LAYOUT_STEREO;
-        }
-        remap->nchannels = av_get_channel_layout_nb_channels(channel_layout);
-
         // build the table and check whether remapping is necessary
         hb_audio_remap_build_table(remap->channel_map_out,
                                    remap->channel_map_in, channel_layout,
diff --git a/libhb/audio_remap.h b/libhb/audio_remap.h
index 3092fb854..6f44ad6b9 100644
--- a/libhb/audio_remap.h
+++ b/libhb/audio_remap.h
@@ -73,8 +73,7 @@ hb_audio_remap_t* hb_audio_remap_init(enum AVSampleFormat sample_fmt,
  * Must be called at least once before remapping.
  */
 void              hb_audio_remap_set_channel_layout(hb_audio_remap_t *remap,
-                                                    uint64_t channel_layout,
-                                                    int channels);
+                                                    uint64_t channel_layout);
 
 /*
  * Free an hb_audio_remap_t.
diff --git a/libhb/audio_resample.c b/libhb/audio_resample.c
index b48cc7271..913186cc0 100644
--- a/libhb/audio_resample.c
+++ b/libhb/audio_resample.c
@@ -75,12 +75,10 @@ fail:
 }
 
 void hb_audio_resample_set_channel_layout(hb_audio_resample_t *resample,
-                                          uint64_t channel_layout,
-                                          int channels)
+                                          uint64_t channel_layout)
 {
     if (resample != NULL)
     {
-        channel_layout = hb_ff_layout_xlat(channel_layout, channels);
         if (channel_layout == AV_CH_LAYOUT_STEREO_DOWNMIX)
         {
             // Dolby Surround is Stereo when it comes to remixing
diff --git a/libhb/audio_resample.h b/libhb/audio_resample.h
index bac3b3292..9242d78a5 100644
--- a/libhb/audio_resample.h
+++ b/libhb/audio_resample.h
@@ -74,13 +74,10 @@ hb_audio_resample_t* hb_audio_resample_init(enum AVSampleFormat sample_fmt,
  *
  * They should be called whenever the relevant characteristic(s) differ from the
  * requested output characteristics, or if they may have changed in the source.
- *
- * Note: channel_layout is automatically sanitized.
  */
 
 void                 hb_audio_resample_set_channel_layout(hb_audio_resample_t *resample,
-                                                          uint64_t channel_layout,
-                                                          int channels);
+                                                          uint64_t channel_layout);
 
 void                 hb_audio_resample_set_mix_levels(hb_audio_resample_t *resample,
                                                       double surround_mix_level,
diff --git a/libhb/deca52.c b/libhb/deca52.c
index f5dac1f96..5dde0378a 100644
--- a/libhb/deca52.c
+++ b/libhb/deca52.c
@@ -38,7 +38,6 @@ struct hb_work_private_s
     uint8_t       buf[6][6][256 * sizeof(float)]; // decoded frame (up to 6 channels, 6 blocks * 256 samples)
     uint8_t      *samples[6];                     // pointers to the start of each plane (1 per channel)
 
-    int                 nchannels;
     int                 use_mix_levels;
     uint64_t            channel_layout;
     hb_audio_remap_t    *remap;
@@ -383,13 +382,10 @@ static hb_buffer_t* Decode(hb_work_object_t *w)
         if (new_layout != pv->channel_layout)
         {
             pv->channel_layout = new_layout;
-            pv->nchannels      = av_get_channel_layout_nb_channels(new_layout);
             hb_audio_remap_set_channel_layout(pv->remap,
-                                              pv->channel_layout,
-                                              pv->nchannels);
+                                              pv->channel_layout);
             hb_audio_resample_set_channel_layout(pv->resample,
-                                                 pv->channel_layout,
-                                                 pv->nchannels);
+                                                 pv->channel_layout);
         }
         if (pv->use_mix_levels)
         {
@@ -416,7 +412,8 @@ static hb_buffer_t* Decode(hb_work_object_t *w)
              *
              * copy samples to our internal buffer
              */
-            for (j = 0; j < pv->nchannels; j++)
+            for (j = 0;
+                 j < av_get_channel_layout_nb_channels(pv->channel_layout); j++)
             {
                 pv->samples[j] = (uint8_t*)pv->buf[j];
                 memcpy(pv->buf[j][i], block_samples, 256 * sizeof(float));
diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c
index a0f5a59c4..5a4b0513c 100644
--- a/libhb/decavcodec.c
+++ b/libhb/decavcodec.c
@@ -655,18 +655,17 @@ static int decavcodecaBSInfo( hb_work_object_t *w, const hb_buffer_t *buf,
                 avp.size = pbuffer_size;
 
                 len = avcodec_decode_audio4(context, frame, &got_frame, &avp);
-                if (len > 0 && context->sample_rate > 0)
+                if (len > 0 && got_frame)
                 {
                     info->rate_base          = 1;
-                    info->rate               = context->sample_rate;
+                    info->rate               = frame->sample_rate;
                     info->samples_per_frame  = frame->nb_samples;
 
                     int bps = av_get_bits_per_sample(context->codec_id);
-                    if (bps > 0 && context->channels > 0)
+                    int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
+                    if (bps > 0)
                     {
-                        info->bitrate = (bps *
-                                         context->channels *
-                                         context->sample_rate);
+                        info->bitrate = (bps * channels * info->rate);
                     }
                     else if (context->bit_rate > 0)
                     {
@@ -683,9 +682,7 @@ static int decavcodecaBSInfo( hb_work_object_t *w, const hb_buffer_t *buf,
                     }
                     else
                     {
-                        info->channel_layout =
-                            hb_ff_layout_xlat(context->channel_layout,
-                                              context->channels);
+                        info->channel_layout = frame->channel_layout;
                     }
 
                     ret = 1;
@@ -2081,26 +2078,11 @@ static void decodeAudio(hb_audio_t *audio, hb_work_private_t *pv, uint8_t *data,
 
         pos += len;
 
-        // set the duration on every frame since the stream format can
-        // change (it shouldn't but there's no way to guarantee it).
-        // duration is a scaling factor to go from #bytes in the decoded
-        // frame to frame time (in 90KHz mpeg ticks). 'channels' converts
-        // total samples to per-channel samples. 'sample_rate' converts
-        // per-channel samples to seconds per sample and the 90000
-        // is mpeg ticks per second.
-        //
-        // Also, sample rate is not available until we have decoded some
-        // audio.
-        if ( pv->context->sample_rate )
-        {
-            pv->duration = 90000. / (double)( pv->context->sample_rate );
-        }
-
         if (got_frame)
         {
             hb_buffer_t *out;
-            double duration = pv->frame->nb_samples * pv->duration;
-            double pts_next = pv->pts_next + duration;
+            double duration = (90000. * pv->frame->nb_samples /
+                               (double)pv->frame->sample_rate);
 
             if (audio->config.out.codec & HB_ACODEC_PASS_FLAG)
             {
@@ -2113,10 +2095,9 @@ static void decodeAudio(hb_audio_t *audio, hb_work_private_t *pv, uint8_t *data,
             else
             {
                 hb_audio_resample_set_channel_layout(pv->resample,
-                                                     context->channel_layout,
-                                                     context->channels);
+                                                     pv->frame->channel_layout);
                 hb_audio_resample_set_sample_fmt(pv->resample,
-                                                 context->sample_fmt);
+                                                 pv->frame->format);
                 if (hb_audio_resample_update(pv->resample))
                 {
                     hb_log("decavcodec: hb_audio_resample_update() failed");
@@ -2132,8 +2113,8 @@ static void decodeAudio(hb_audio_t *audio, hb_work_private_t *pv, uint8_t *data,
             {
                 out->s.start    = pv->pts_next;
                 out->s.duration = duration;
-                out->s.stop     = pts_next;
-                pv->pts_next    = pts_next;
+                out->s.stop     = duration + pv->pts_next;
+                pv->pts_next    = duration + pv->pts_next;
                 hb_list_add(pv->list, out);
             }
         }
diff --git a/libhb/declpcm.c b/libhb/declpcm.c
index 7a17f2a14..a8f2e32d4 100644
--- a/libhb/declpcm.c
+++ b/libhb/declpcm.c
@@ -330,8 +330,7 @@ static hb_buffer_t *Decode( hb_work_object_t *w )
     }
 
     hb_audio_resample_set_channel_layout(pv->resample,
-                                         hdr2layout[pv->nchannels - 1],
-                                         pv->nchannels);
+                                         hdr2layout[pv->nchannels - 1]);
     if (hb_audio_resample_update(pv->resample))
     {
         hb_log("declpcm: hb_audio_resample_update() failed");
diff --git a/libhb/hb.c b/libhb/hb.c
index 2b9609aa6..156d1241d 100644
--- a/libhb/hb.c
+++ b/libhb/hb.c
@@ -288,23 +288,6 @@ uint64_t hb_ff_mixdown_xlat(int hb_mixdown, int *downmix_mode)
     return ff_layout;
 }
 
-uint64_t hb_ff_layout_xlat(uint64_t ff_channel_layout, int nchannels)
-{
-    uint64_t hb_layout = ff_channel_layout;
-    if (!hb_layout || av_get_channel_layout_nb_channels(hb_layout) != nchannels)
-    {
-        hb_layout = av_get_default_channel_layout(nchannels);
-        if (!hb_layout)
-        {
-            // This will likely not sound very good ;)
-            hb_layout = AV_CH_LAYOUT_STEREO;
-            hb_error("hb_ff_layout_xlat: unsupported layout 0x%"PRIx64" with %d channels",
-                     ff_channel_layout, nchannels);
-        }
-    }
-    return hb_layout;
-}
-
 /*
  * Set sample format to the request format if supported by the codec.
  * The planar/packed variant of the requested format is the next best thing.
diff --git a/libhb/hbffmpeg.h b/libhb/hbffmpeg.h
index 776eec61a..94b1a903e 100644
--- a/libhb/hbffmpeg.h
+++ b/libhb/hbffmpeg.h
@@ -23,7 +23,6 @@ void hb_avcodec_init(void);
 int  hb_avcodec_open(AVCodecContext *, AVCodec *, AVDictionary **, int);
 int  hb_avcodec_close(AVCodecContext *);
 
-uint64_t hb_ff_layout_xlat(uint64_t ff_channel_layout, int nchannels);
 uint64_t hb_ff_mixdown_xlat(int hb_mixdown, int *downmix_mode);
 void     hb_ff_set_sample_fmt(AVCodecContext *, AVCodec *, enum AVSampleFormat);
 
diff --git a/libhb/platform/macosx/encca_aac.c b/libhb/platform/macosx/encca_aac.c
index 19e182a22..882b97a5a 100644
--- a/libhb/platform/macosx/encca_aac.c
+++ b/libhb/platform/macosx/encca_aac.c
@@ -297,7 +297,7 @@ int encCoreAudioInit(hb_work_object_t *w, hb_job_t *job, enum AAC_MODE mode)
         hb_error("encCoreAudioInit: hb_audio_remap_init() failed");
     }
     uint64_t layout = hb_ff_mixdown_xlat(audio->config.out.mixdown, NULL);
-    hb_audio_remap_set_channel_layout(pv->remap, layout, pv->nchannels);
+    hb_audio_remap_set_channel_layout(pv->remap, layout);
 
     // get maximum output size
     AudioConverterGetProperty(pv->converter,