aout_buffer_t.start_data -> aout_buffer_t.i_pts

This will be needed to replace aout_buffer_t with block_t.
We also need to replace end_date with i_length, which is a bit more
involved.

include/vlc_aout.h

@@ -134,7 +134,7 @@ struct aout_buffer_t
     size_t                  i_size, i_nb_bytes;
     unsigned int            i_nb_samples;
     bool                    b_discontinuity; /* Set on discontinuity (for non pcm stream) */
-    mtime_t                 start_date, end_date;
+    mtime_t                 i_pts, end_date;
 
     struct aout_buffer_t *  p_next;
     void                 *p_sys;

modules/audio_filter/converter/dtstospdif.c

@@ -146,7 +146,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
     {
         if( p_filter->p_sys->i_frames == 1 )
             /* We'll need the starting date */
-            p_filter->p_sys->start_date = p_in_buf->start_date;
+            p_filter->p_sys->start_date = p_in_buf->i_pts;
 
         /* Not enough data */
         p_out_buf->i_nb_samples = 0;
@@ -213,7 +213,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
         }
     }
 
-    p_out_buf->start_date = p_filter->p_sys->start_date;
+    p_out_buf->i_pts = p_filter->p_sys->start_date;
     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples * 3;
     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples * 4;
 }

modules/audio_filter/resampler/bandlimited.c

@@ -219,7 +219,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
             p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
                 p_sys->i_old_wing;
 
-            p_out_buf->start_date = date_Get( &p_sys->end_date );
+            p_out_buf->i_pts = date_Get( &p_sys->end_date );
             p_out_buf->end_date =
                 date_Increment( &p_sys->end_date,
                                 p_out_buf->i_nb_samples );
@@ -239,7 +239,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
         p_filter->b_continuity = true;
         p_sys->i_remainder = 0;
         date_Init( &p_sys->end_date, i_out_rate, 1 );
-        date_Set( &p_sys->end_date, p_in_buf->start_date );
+        date_Set( &p_sys->end_date, p_in_buf->i_pts );
         p_sys->i_old_rate   = p_filter->input.i_rate;
         p_sys->d_old_factor = 1;
         p_sys->i_old_wing   = 0;
@@ -454,7 +454,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 
     /* Finalize aout buffer */
     p_out_buf->i_nb_samples = i_out;
-    p_out_buf->start_date = date_Get( &p_sys->end_date );
+    p_out_buf->i_pts = date_Get( &p_sys->end_date );
     p_out_buf->end_date = date_Increment( &p_sys->end_date,
                                           p_out_buf->i_nb_samples );
 

modules/audio_filter/resampler/linear.c

@@ -223,12 +223,12 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
     }
 
     p_out_buf->i_nb_samples = i_out;
-    p_out_buf->start_date = p_in_buf->start_date;
+    p_out_buf->i_pts = p_in_buf->i_pts;
 
-    if( p_in_buf->start_date !=
+    if( p_in_buf->i_pts !=
         date_Get( &p_sys->end_date ) )
     {
-        date_Set( &p_sys->end_date, p_in_buf->start_date );
+        date_Set( &p_sys->end_date, p_in_buf->i_pts );
     }
 
     p_out_buf->end_date = date_Increment( &p_sys->end_date,

modules/audio_filter/resampler/trivial.c

@@ -112,7 +112,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 
     p_out_buf->i_nb_samples = i_out_nb;
     p_out_buf->i_nb_bytes = i_out_nb * i_sample_bytes;
-    p_out_buf->start_date = p_in_buf->start_date;
-    p_out_buf->end_date = p_out_buf->start_date + p_out_buf->i_nb_samples *
+    p_out_buf->i_pts = p_in_buf->i_pts;
+    p_out_buf->end_date = p_out_buf->i_pts + p_out_buf->i_nb_samples *
         1000000 / p_filter->output.i_rate;
 }

modules/audio_filter/resampler/ugly.c

@@ -125,7 +125,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 
     p_out_buf->i_nb_samples = i_out_nb;
     p_out_buf->i_nb_bytes = i_out_nb * i_sample_bytes;
-    p_out_buf->start_date = p_in_buf->start_date;
-    p_out_buf->end_date = p_out_buf->start_date + p_out_buf->i_nb_samples *
+    p_out_buf->i_pts = p_in_buf->i_pts;
+    p_out_buf->end_date = p_out_buf->i_pts + p_out_buf->i_nb_samples *
         1000000 / p_filter->output.i_rate;
 }

modules/audio_output/oss.c

@@ -663,7 +663,7 @@ static void* OSSThread( vlc_object_t *p_this )
             i_size = p_buffer->i_nb_bytes;
             /* This is theoretical ... we'll see next iteration whether
              * we're drifting */
-            next_date += p_buffer->end_date - p_buffer->start_date;
+            next_date += p_buffer->end_date - p_buffer->i_pts;
         }
         else
         {

modules/audio_output/waveout.c

@@ -1063,7 +1063,7 @@ static void* WaveOutThread( vlc_object_t *p_this )
                 if( p_buffer )
                 {
                     mtime_t buffer_length = (p_buffer->end_date
-                                             - p_buffer->start_date);
+                                             - p_buffer->i_pts);
                     next_date = next_date + buffer_length;
                     i_buffer_length = buffer_length/1000;
                 }

modules/codec/a52.c

@@ -393,7 +393,7 @@ static aout_buffer_t *GetAoutBuffer( decoder_t *p_dec )
     p_buf = decoder_NewAudioBuffer( p_dec, p_sys->frame.i_samples );
     if( p_buf == NULL ) return NULL;
 
-    p_buf->start_date = date_Get( &p_sys->end_date );
+    p_buf->i_pts = date_Get( &p_sys->end_date );
     p_buf->end_date = date_Increment( &p_sys->end_date, p_sys->frame.i_samples );
 
     return p_buf;

modules/codec/adpcm.c

@@ -300,7 +300,7 @@ static aout_buffer_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
             return NULL;
         }
 
-        p_out->start_date = date_Get( &p_sys->end_date );
+        p_out->i_pts = date_Get( &p_sys->end_date );
         p_out->end_date =
             date_Increment( &p_sys->end_date, p_sys->i_samplesperblock );
 

modules/codec/aes3.c

@@ -134,7 +134,7 @@ static aout_buffer_t *Decode( decoder_t *p_dec, block_t **pp_block )
     if( p_aout_buffer == NULL )
         goto exit;
 
-    p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+    p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
     p_aout_buffer->end_date = date_Increment( &p_sys->end_date, i_frame_length );
 
     p_block->i_buffer -= AES3_HEADER_LEN;

modules/codec/araw.c

@@ -370,7 +370,7 @@ static aout_buffer_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
         return NULL;
     }
 
-    p_out->start_date = date_Get( &p_sys->end_date );
+    p_out->i_pts = date_Get( &p_sys->end_date );
     p_out->end_date   = date_Increment( &p_sys->end_date, i_samples );
 
     if( p_sys->p_logtos16 )
@@ -1419,7 +1419,7 @@ static block_t *EncoderEncode( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
 
     if( p_block )
     {
-        p_block->i_dts = p_block->i_pts = p_aout_buf->start_date;
+        p_block->i_dts = p_block->i_pts = p_aout_buf->i_pts;
         p_block->i_length = (int64_t)p_aout_buf->i_nb_samples *
             (int64_t)1000000 / p_enc->fmt_in.audio.i_rate;
     }

modules/codec/avcodec/audio.c

@@ -239,7 +239,7 @@ static aout_buffer_t *SplitBuffer( decoder_t *p_dec )
     if( ( p_buffer = decoder_NewAudioBuffer( p_dec, i_samples ) ) == NULL )
         return NULL;
 
-    p_buffer->start_date = date_Get( &p_sys->end_date );
+    p_buffer->i_pts = date_Get( &p_sys->end_date );
     p_buffer->end_date = date_Increment( &p_sys->end_date, i_samples );
 
     if( p_sys->b_extract )

modules/codec/avcodec/encoder.c

@@ -1055,7 +1055,7 @@ static block_t *EncodeAudio( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
     int i_samples = p_aout_buf->i_nb_samples;
     int i_samples_delay = p_sys->i_samples_delay;
 
-    p_sys->i_pts = p_aout_buf->start_date -
+    p_sys->i_pts = p_aout_buf->i_pts -
                 (mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
                 (mtime_t)p_enc->fmt_in.audio.i_rate;
 

modules/codec/dmo/dmo.c

@@ -981,7 +981,7 @@ static void *DecBlock( decoder_t *p_dec, block_t **pp_block )
             memcpy( p_aout_buffer->p_buffer,
                     block_out.p_buffer, block_out.i_buffer );
             /* Date management */
-            p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+            p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
             p_aout_buffer->end_date =
                 date_Increment( &p_sys->end_date, i_samples );
         }
@@ -1538,7 +1538,7 @@ static block_t *EncodeBlock( encoder_t *p_enc, void *p_data )
         memcpy( p_block_in->p_buffer, p_aout_buffer->p_buffer,
                 p_block_in->i_buffer );
 
-        i_pts = p_aout_buffer->start_date;
+        i_pts = p_aout_buffer->i_pts;
     }
 
     /* Feed input to the DMO */

modules/codec/dts.c

@@ -410,7 +410,7 @@ static aout_buffer_t *GetAoutBuffer( decoder_t *p_dec )
     p_buf->i_nb_samples = p_sys->i_frame_length;
     p_buf->i_nb_bytes = p_sys->i_frame_size;
 
-    p_buf->start_date = date_Get( &p_sys->end_date );
+    p_buf->i_pts = date_Get( &p_sys->end_date );
     p_buf->end_date =
         date_Increment( &p_sys->end_date, p_sys->i_frame_length );
 

modules/codec/faad.c

@@ -414,7 +414,7 @@ static aout_buffer_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
             return NULL;
         }
 
-        p_out->start_date = date_Get( &p_sys->date );
+        p_out->i_pts = date_Get( &p_sys->date );
         p_out->end_date = date_Increment( &p_sys->date, frame.samples / frame.channels );
 
         DoReordering( (uint32_t *)p_out->p_buffer, samples,

modules/codec/flac.c

@@ -662,7 +662,7 @@ DecoderWriteCallback( const FLAC__StreamDecoder *decoder,
     }
 
     /* Date management (already done by packetizer) */
-    p_sys->p_aout_buffer->start_date = p_sys->p_block->i_pts;
+    p_sys->p_aout_buffer->i_pts = p_sys->p_block->i_pts;
     p_sys->p_aout_buffer->end_date =
         p_sys->p_block->i_pts + p_sys->p_block->i_length;
 
@@ -1359,7 +1359,7 @@ static block_t *Encode( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
     block_t *p_chain;
     unsigned int i;
 
-    p_sys->i_pts = p_aout_buf->start_date -
+    p_sys->i_pts = p_aout_buf->i_pts -
                 (mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
                 (mtime_t)p_enc->fmt_in.audio.i_rate;
 

modules/codec/fluidsynth.c

@@ -197,7 +197,7 @@ static aout_buffer_t *DecodeBlock (decoder_t *p_dec, block_t **pp_block)
     if (p_out == NULL)
         goto drop;
 
-    p_out->start_date = date_Get (&p_sys->end_date );
+    p_out->i_pts = date_Get (&p_sys->end_date );
     p_out->end_date   = date_Increment (&p_sys->end_date, samples);
     if (!p_sys->fixed)
         fluid_synth_write_float (p_sys->synth, samples,

modules/codec/lpcm.c

@@ -360,7 +360,7 @@ static void *DecodeFrame( decoder_t *p_dec, block_t **pp_block )
         if( !p_aout_buffer )
             return NULL;
 
-        p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+        p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
         p_aout_buffer->end_date =
             date_Increment( &p_sys->end_date, i_frame_length );
 

modules/codec/mpeg_audio.c

@@ -538,7 +538,7 @@ static aout_buffer_t *GetAoutBuffer( decoder_t *p_dec )
     p_buf = decoder_NewAudioBuffer( p_dec, p_sys->i_frame_length );
     if( p_buf == NULL ) return NULL;
 
-    p_buf->start_date = date_Get( &p_sys->end_date );
+    p_buf->i_pts = date_Get( &p_sys->end_date );
     p_buf->end_date =
         date_Increment( &p_sys->end_date, p_sys->i_frame_length );
     p_buf->b_discontinuity = p_sys->b_discontinuity;

modules/codec/quicktime.c

@@ -646,7 +646,7 @@ static aout_buffer_t *DecodeAudio( decoder_t *p_dec, block_t **pp_block )
 
         if( p_out )
         {
-            p_out->start_date = date_Get( &p_sys->date );
+            p_out->i_pts = date_Get( &p_sys->date );
             p_out->end_date = date_Increment( &p_sys->date, i_frames );
 
             memcpy( p_out->p_buffer,

modules/codec/realaudio.c

@@ -723,7 +723,7 @@ static aout_buffer_t *Decode( decoder_t *p_dec, block_t **pp_block )
         memcpy( p_aout_buffer->p_buffer, p_sys->p_out, p_sys->i_out );
 
         /* Date management */
-        p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+        p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
         p_aout_buffer->end_date =
             date_Increment( &p_sys->end_date, i_samples );
     }

modules/codec/shine/shine_mod.c

@@ -184,7 +184,7 @@ static block_t *EncodeFrame( encoder_t *p_enc, aout_buffer_t *p_block )
     block_t *p_pcm_block;
     block_t *p_chain = NULL;
     unsigned int i_samples = p_block->i_nb_bytes >> 2 /* s16l stereo */;
-    mtime_t start_date = p_block->start_date;
+    mtime_t start_date = p_block->i_pts;
     start_date -= (mtime_t)i_samples * (mtime_t)1000000 / (mtime_t)p_enc->fmt_out.audio.i_rate;
 
     VLC_UNUSED(p_enc);

modules/codec/speex.c

@@ -711,7 +711,7 @@ static aout_buffer_t *DecodeRtpSpeexPacket( decoder_t *p_dec, block_t **pp_block
     /* 
       Handle date management on the audio output buffer. 
     */
-    p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+    p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
     p_aout_buffer->end_date = date_Increment( &p_sys->end_date, 
         p_sys->p_header->frame_size );
     
@@ -771,7 +771,7 @@ static aout_buffer_t *DecodePacket( decoder_t *p_dec, ogg_packet *p_oggpacket )
                                      &p_sys->stereo );
 
         /* Date management */
-        p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+        p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
         p_aout_buffer->end_date =
             date_Increment( &p_sys->end_date, p_sys->p_header->frame_size );
 
@@ -1037,7 +1037,7 @@ static block_t *Encode( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
     int i_samples = p_aout_buf->i_nb_samples;
     int i_samples_delay = p_sys->i_samples_delay;
 
-    p_sys->i_pts = p_aout_buf->start_date -
+    p_sys->i_pts = p_aout_buf->i_pts -
                 (mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
                 (mtime_t)p_enc->fmt_in.audio.i_rate;
 

modules/codec/twolame.c

@@ -271,7 +271,7 @@ static block_t *Encode( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
     int i_nb_samples = p_aout_buf->i_nb_samples;
     block_t *p_chain = NULL;
 
-    p_sys->i_pts = p_aout_buf->start_date -
+    p_sys->i_pts = p_aout_buf->i_pts -
                 (mtime_t)1000000 * (mtime_t)p_sys->i_nb_samples /
                 (mtime_t)p_enc->fmt_out.audio.i_rate;
 

modules/codec/vorbis.c

@@ -559,7 +559,7 @@ static aout_buffer_t *DecodePacket( decoder_t *p_dec, ogg_packet *p_oggpacket )
         vorbis_synthesis_read( &p_sys->vd, i_samples );
 
         /* Date management */
-        p_aout_buffer->start_date = date_Get( &p_sys->end_date );
+        p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
         p_aout_buffer->end_date = date_Increment( &p_sys->end_date, i_samples );
         return p_aout_buffer;
     }
@@ -915,7 +915,7 @@ static block_t *Encode( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
     int i;
     unsigned int j;
 
-    p_sys->i_pts = p_aout_buf->start_date -
+    p_sys->i_pts = p_aout_buf->i_pts -
                 (mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
                 (mtime_t)p_enc->fmt_in.audio.i_rate;
 

modules/codec/wmafixed/wma.c

@@ -101,7 +101,7 @@ static aout_buffer_t *SplitBuffer( decoder_t *p_dec )
     if( !( p_buffer = p_dec->pf_aout_buffer_new( p_dec, i_samples ) ) )
         return NULL;
 
-    p_buffer->start_date = date_Get( &p_sys->end_date );
+    p_buffer->i_pts = date_Get( &p_sys->end_date );
     p_buffer->end_date = date_Increment( &p_sys->end_date, i_samples );
 
     memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_nb_bytes );

modules/stream_out/transcode.c

@@ -1295,25 +1295,25 @@ static int transcode_audio_process( sout_stream_t *p_stream,
         if( p_sys->b_master_sync )
         {
             mtime_t i_dts = date_Get( &id->interpolated_pts ) + 1;
-            if ( p_audio_buf->start_date - i_dts > MASTER_SYNC_MAX_DRIFT
-                  || p_audio_buf->start_date - i_dts < -MASTER_SYNC_MAX_DRIFT )
+            if ( p_audio_buf->i_pts - i_dts > MASTER_SYNC_MAX_DRIFT
+                  || p_audio_buf->i_pts - i_dts < -MASTER_SYNC_MAX_DRIFT )
             {
                 msg_Dbg( p_stream, "drift is too high, resetting master sync" );
-                date_Set( &id->interpolated_pts, p_audio_buf->start_date );
-                i_dts = p_audio_buf->start_date + 1;
+                date_Set( &id->interpolated_pts, p_audio_buf->i_pts );
+                i_dts = p_audio_buf->i_pts + 1;
             }
-            p_sys->i_master_drift = p_audio_buf->start_date - i_dts;
+            p_sys->i_master_drift = p_audio_buf->i_pts - i_dts;
             date_Increment( &id->interpolated_pts, p_audio_buf->i_nb_samples );
-            p_audio_buf->start_date -= p_sys->i_master_drift;
+            p_audio_buf->i_pts -= p_sys->i_master_drift;
             p_audio_buf->end_date -= p_sys->i_master_drift;
         }
 
         p_audio_block = p_audio_buf->p_sys;
         p_audio_block->i_buffer = p_audio_buf->i_nb_bytes;
         p_audio_block->i_dts = p_audio_block->i_pts =
-            p_audio_buf->start_date;
+            p_audio_buf->i_pts;
         p_audio_block->i_length = p_audio_buf->end_date -
-            p_audio_buf->start_date;
+            p_audio_buf->i_pts;
         p_audio_block->i_samples = p_audio_buf->i_nb_samples;
 
         /* Run filter chain */
@@ -1329,7 +1329,7 @@ static int transcode_audio_process( sout_stream_t *p_stream,
         p_audio_buf->p_buffer = p_audio_block->p_buffer;
         p_audio_buf->i_nb_bytes = p_audio_block->i_buffer;
         p_audio_buf->i_nb_samples = p_audio_block->i_samples;
-        p_audio_buf->start_date = p_audio_block->i_dts;
+        p_audio_buf->i_pts = p_audio_block->i_dts;
         p_audio_buf->end_date = p_audio_block->i_dts + p_audio_block->i_length;
 
         audio_timer_start( id->p_encoder );

modules/visualization/goom.c

@@ -238,7 +238,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
         return;
     }
     memcpy( p_block->p_buffer, p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
-    p_block->i_pts = p_in_buf->start_date;
+    p_block->i_pts = p_in_buf->i_pts;
 
     p_sys->p_thread->pp_blocks[p_sys->p_thread->i_blocks++] = p_block;
 

modules/visualization/visual/visual.c

@@ -375,7 +375,7 @@ static void DoWork( aout_instance_t *p_aout, aout_filter_t *p_filter,
 #undef p_effect
     }
 
-    p_outpic->date = ( p_in_buf->start_date + p_in_buf->end_date ) / 2;
+    p_outpic->date = ( p_in_buf->i_pts + p_in_buf->end_date ) / 2;
 
     vout_DisplayPicture( p_sys->p_vout, p_outpic );
 }

src/audio_output/common.c

@@ -369,7 +369,7 @@ void aout_FifoPush( aout_instance_t * p_aout, aout_fifo_t * p_fifo,
     /* Enforce the continuity of the stream. */
     if ( date_Get( &p_fifo->end_date ) )
     {
-        p_buffer->start_date = date_Get( &p_fifo->end_date );
+        p_buffer->i_pts = date_Get( &p_fifo->end_date );
         p_buffer->end_date = date_Increment( &p_fifo->end_date,
                                              p_buffer->i_nb_samples );
     }
@@ -416,7 +416,7 @@ void aout_FifoMoveDates( aout_instance_t * p_aout, aout_fifo_t * p_fifo,
     p_buffer = p_fifo->p_first;
     while ( p_buffer != NULL )
     {
-        p_buffer->start_date += difference;
+        p_buffer->i_pts += difference;
         p_buffer->end_date += difference;
         p_buffer = p_buffer->p_next;
     }
@@ -440,7 +440,7 @@ mtime_t aout_FifoFirstDate( aout_instance_t * p_aout, aout_fifo_t * p_fifo )
 {
     (void)p_aout;
     AOUT_ASSERT_FIFO_LOCKED;
-    return p_fifo->p_first ?  p_fifo->p_first->start_date : 0;
+    return p_fifo->p_first ?  p_fifo->p_first->i_pts : 0;
 }
 
 /*****************************************************************************

src/audio_output/dec.c

@@ -281,7 +281,7 @@ aout_buffer_t * aout_DecNewBuffer( aout_input_t * p_input,
         return NULL;
 
     p_buffer->i_nb_samples = i_nb_samples;
-    p_buffer->start_date = p_buffer->end_date = 0;
+    p_buffer->i_pts = p_buffer->end_date = 0;
     return p_buffer;
 }
 
@@ -304,9 +304,9 @@ int aout_DecPlay( aout_instance_t * p_aout, aout_input_t * p_input,
     assert( i_input_rate >= INPUT_RATE_DEFAULT / AOUT_MAX_INPUT_RATE &&
             i_input_rate <= INPUT_RATE_DEFAULT * AOUT_MAX_INPUT_RATE );
 
-    assert( p_buffer->start_date > 0 );
+    assert( p_buffer->i_pts > 0 );
 
-    p_buffer->end_date = p_buffer->start_date
+    p_buffer->end_date = p_buffer->i_pts
                             + (mtime_t)p_buffer->i_nb_samples * 1000000
                                 / p_input->input.i_rate;
 
@@ -331,7 +331,7 @@ int aout_DecPlay( aout_instance_t * p_aout, aout_input_t * p_input,
                     p_buffer->i_nb_bytes );
         p_new_buffer->i_nb_samples = p_buffer->i_nb_samples;
         p_new_buffer->i_nb_bytes = p_buffer->i_nb_bytes;
-        p_new_buffer->start_date = p_buffer->start_date;
+        p_new_buffer->i_pts = p_buffer->i_pts;
         p_new_buffer->end_date = p_buffer->end_date;
         aout_BufferFree( p_buffer );
         p_buffer = p_new_buffer;
@@ -383,7 +383,7 @@ void aout_DecChangePause( aout_instance_t *p_aout, aout_input_t *p_input, bool b
         aout_lock_mixer( p_aout );
         for( aout_buffer_t *p = p_input->mixer.fifo.p_first; p != NULL; p = p->p_next )
         {
-            p->start_date += i_duration;
+            p->i_pts += i_duration;
             p->end_date += i_duration;
         }
         aout_unlock_mixer( p_aout );

src/audio_output/filters.c

@@ -353,7 +353,7 @@ void aout_FiltersPlay( aout_instance_t * p_aout,
                               *pp_input_buffer );
         if( p_output_buffer == NULL )
             return;
-        p_output_buffer->start_date = (*pp_input_buffer)->start_date;
+        p_output_buffer->i_pts = (*pp_input_buffer)->i_pts;
         p_output_buffer->end_date = (*pp_input_buffer)->end_date;
 
         /* Please note that p_output_buffer->i_nb_samples & i_nb_bytes

src/audio_output/input.c

@@ -552,7 +552,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
     /* Actually run the resampler now. */
     if ( p_input->i_nb_resamplers > 0 )
     {
-        const mtime_t i_date = p_buffer->start_date;
+        const mtime_t i_date = p_buffer->i_pts;
         aout_FiltersPlay( p_aout, p_input->pp_resamplers,
                           p_input->i_nb_resamplers,
                           &p_buffer );
@@ -600,12 +600,12 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
         start_date = 0;
     }
 
-    if ( p_buffer->start_date < mdate() + AOUT_MIN_PREPARE_TIME )
+    if ( p_buffer->i_pts < mdate() + AOUT_MIN_PREPARE_TIME )
     {
         /* The decoder gives us f*cked up PTS. It's its business, but we
          * can't present it anyway, so drop the buffer. */
         msg_Warn( p_aout, "PTS is out of range (%"PRId64"), dropping buffer",
-                  mdate() - p_buffer->start_date );
+                  mdate() - p_buffer->i_pts );
 
         inputDrop( p_input, p_buffer );
         inputResamplingStop( p_input );
@@ -616,10 +616,10 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
      * the audio. */
     mtime_t i_pts_tolerance = 3 * AOUT_PTS_TOLERANCE * i_input_rate / INPUT_RATE_DEFAULT;
     if ( start_date != 0 &&
-         ( start_date < p_buffer->start_date - i_pts_tolerance ) )
+         ( start_date < p_buffer->i_pts - i_pts_tolerance ) )
     {
         msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out",
-                  start_date - p_buffer->start_date );
+                  start_date - p_buffer->i_pts );
         aout_lock_input_fifos( p_aout );
         aout_FifoSet( p_aout, &p_input->mixer.fifo, 0 );
         p_input->mixer.begin = NULL;
@@ -630,15 +630,15 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
         start_date = 0;
     }
     else if ( start_date != 0 &&
-              ( start_date > p_buffer->start_date + i_pts_tolerance) )
+              ( start_date > p_buffer->i_pts + i_pts_tolerance) )
     {
         msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer",
-                  start_date - p_buffer->start_date );
+                  start_date - p_buffer->i_pts );
         inputDrop( p_input, p_buffer );
         return 0;
     }
 
-    if ( start_date == 0 ) start_date = p_buffer->start_date;
+    if ( start_date == 0 ) start_date = p_buffer->i_pts;
 
 #ifndef AOUT_PROCESS_BEFORE_CHEKS
     /* Run pre-filters. */
@@ -649,8 +649,8 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
     /* Run the resampler if needed.
      * We first need to calculate the output rate of this resampler. */
     if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) &&
-         ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
-           || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) &&
+         ( start_date < p_buffer->i_pts - AOUT_PTS_TOLERANCE
+           || start_date > p_buffer->i_pts + AOUT_PTS_TOLERANCE ) &&
          p_input->i_nb_resamplers > 0 )
     {
         /* Can happen in several circumstances :
@@ -660,7 +660,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
          *    synchronization
          * Solution : resample the buffer to avoid a scratch.
          */
-        mtime_t drift = p_buffer->start_date - start_date;
+        mtime_t drift = p_buffer->i_pts - start_date;
 
         p_input->i_resamp_start_date = mdate();
         p_input->i_resamp_start_drift = (int)drift;
@@ -703,9 +703,9 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
             msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec "
                       "(drift: %"PRIi64")",
                       mdate() - p_input->i_resamp_start_date,
-                      p_buffer->start_date - start_date);
+                      p_buffer->i_pts - start_date);
         }
-        else if( abs( (int)(p_buffer->start_date - start_date) ) <
+        else if( abs( (int)(p_buffer->i_pts - start_date) ) <
                  abs( p_input->i_resamp_start_drift ) / 2 )
         {
             /* if we reduced the drift from half, then it is time to switch
@@ -717,7 +717,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
             p_input->i_resamp_start_drift = 0;
         }
         else if( p_input->i_resamp_start_drift &&
-                 ( abs( (int)(p_buffer->start_date - start_date) ) >
+                 ( abs( (int)(p_buffer->i_pts - start_date) ) >
                    abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
         {
             /* If the drift is increasing and not decreasing, than something
@@ -745,8 +745,8 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
 
     /* Adding the start date will be managed by aout_FifoPush(). */
     p_buffer->end_date = start_date +
-        (p_buffer->end_date - p_buffer->start_date);
-    p_buffer->start_date = start_date;
+        (p_buffer->end_date - p_buffer->i_pts);
+    p_buffer->i_pts = start_date;
 
     aout_lock_input_fifos( p_aout );
     aout_FifoPush( p_aout, &p_input->mixer.fifo, p_buffer );

src/audio_output/mixer.c

@@ -169,10 +169,10 @@ static int MixBuffer( aout_instance_t * p_aout )
                 continue;
 
             p_buffer = p_fifo->p_first;
-            while ( p_buffer != NULL && p_buffer->start_date < mdate() )
+            while ( p_buffer != NULL && p_buffer->i_pts < mdate() )
             {
                 msg_Warn( p_aout, "input PTS is out of range (%"PRId64"), "
-                          "trashing", mdate() - p_buffer->start_date );
+                          "trashing", mdate() - p_buffer->i_pts );
                 p_buffer = aout_FifoPop( p_aout, p_fifo );
                 aout_BufferFree( p_buffer );
                 p_buffer = p_fifo->p_first;
@@ -184,10 +184,10 @@ static int MixBuffer( aout_instance_t * p_aout )
                 break;
             }
 
-            if ( !start_date || start_date < p_buffer->start_date )
+            if ( !start_date || start_date < p_buffer->i_pts )