* input.c : internal management of input streams for the audio output
*****************************************************************************
* Copyright (C) 2002 VideoLAN
- * $Id: input.c,v 1.15 2002/10/04 18:07:22 sam Exp $
+ * $Id: input.c,v 1.29 2002/12/25 02:23:37 massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
- *
+ *
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
*****************************************************************************/
int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
{
- audio_sample_format_t intermediate_format;
+ audio_sample_format_t intermediate_format, headphone_intermediate_format;
+ aout_filter_t * p_headphone_filter;
+
+ aout_FormatPrint( p_aout, "input", &p_input->input );
/* Prepare FIFO. */
aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate );
/* Create filters. */
memcpy( &intermediate_format, &p_aout->mixer.mixer,
sizeof(audio_sample_format_t) );
+ memcpy( &headphone_intermediate_format, &p_aout->mixer.mixer,
+ sizeof(audio_sample_format_t) );
+ if ( config_GetInt( p_aout , "headphone" ) )
+ {
+ headphone_intermediate_format.i_physical_channels = p_input->input.i_physical_channels;
+ headphone_intermediate_format.i_original_channels = p_input->input.i_original_channels;
+ headphone_intermediate_format.i_bytes_per_frame =
+ headphone_intermediate_format.i_bytes_per_frame
+ * aout_FormatNbChannels( &headphone_intermediate_format )
+ / aout_FormatNbChannels( &intermediate_format );
+ }
+
intermediate_format.i_rate = p_input->input.i_rate;
+ headphone_intermediate_format.i_rate = p_input->input.i_rate;
if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
&p_input->i_nb_filters, &p_input->input,
- &intermediate_format ) < 0 )
+ &headphone_intermediate_format ) < 0 )
{
msg_Err( p_aout, "couldn't set an input pipeline" );
return -1;
}
- /* Create resamplers. */
- intermediate_format.i_rate = (p_input->input.i_rate
- * (100 + AOUT_MAX_RESAMPLING)) / 100;
- if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate )
- {
- /* Just in case... */
- intermediate_format.i_rate++;
- }
- if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
- &p_input->i_nb_resamplers,
- &intermediate_format,
- &p_aout->mixer.mixer ) < 0 )
+ if ( config_GetInt( p_aout , "headphone" ) )
{
- msg_Err( p_aout, "couldn't set a resampler pipeline" );
+ /* create a vlc object */
+ p_headphone_filter = vlc_object_create( p_aout
+ , sizeof(aout_filter_t) );
+ if ( p_headphone_filter == NULL )
+ {
+ msg_Err( p_aout, "couldn't open the headphone virtual spatialization module" );
+ aout_FifoDestroy( p_aout, &p_input->fifo );
+ p_input->b_error = 1;
+ return -1;
+ }
+ vlc_object_attach( p_headphone_filter, p_aout );
+
+ /* find the headphone filter */
+ memcpy( &p_headphone_filter->input, &headphone_intermediate_format
+ , sizeof(audio_sample_format_t) );
+ memcpy( &p_headphone_filter->output, &intermediate_format
+ , sizeof(audio_sample_format_t) );
+ p_headphone_filter->p_module = module_Need( p_headphone_filter, "audio filter"
+ , "headphone" );
+ if ( p_headphone_filter->p_module == NULL )
+ {
+ vlc_object_detach( p_headphone_filter );
+ vlc_object_destroy( p_headphone_filter );
- aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
- p_input->i_nb_filters );
- aout_FifoDestroy( p_aout, &p_input->fifo );
- p_input->b_error = 1;
+ msg_Err( p_aout, "couldn't open the headphone virtual spatialization module" );
+ aout_FifoDestroy( p_aout, &p_input->fifo );
+ p_input->b_error = 1;
+ return -1;
+ }
- return -1;
+ /* success */
+ p_headphone_filter->b_reinit = VLC_TRUE;
+ p_input->pp_filters[p_input->i_nb_filters++] = p_headphone_filter;
}
/* Prepare hints for the buffer allocator. */
p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
p_input->input_alloc.i_bytes_per_sec = -1;
- aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
- p_input->i_nb_resamplers,
- &p_input->input_alloc );
+ if ( AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) )
+ {
+ p_input->i_nb_resamplers = 0;
+ }
+ else
+ {
+ /* Create resamplers. */
+ intermediate_format.i_rate = (__MAX(p_input->input.i_rate,
+ p_aout->mixer.mixer.i_rate)
+ * (100 + AOUT_MAX_RESAMPLING)) / 100;
+ if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate )
+ {
+ /* Just in case... */
+ intermediate_format.i_rate++;
+ }
+ if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
+ &p_input->i_nb_resamplers,
+ &intermediate_format,
+ &p_aout->mixer.mixer ) < 0 )
+ {
+ msg_Err( p_aout, "couldn't set a resampler pipeline" );
+ aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
+ p_input->i_nb_filters );
+ aout_FifoDestroy( p_aout, &p_input->fifo );
+ p_input->b_error = 1;
+
+ return -1;
+ }
+
+ aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
+ p_input->i_nb_resamplers,
+ &p_input->input_alloc );
+
+ /* Setup the initial rate of the resampler */
+ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ }
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+
+ p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
+ p_input->input_alloc.i_bytes_per_sec = -1;
aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
p_input->i_nb_filters,
&p_input->input_alloc );
/* i_bytes_per_sec is still == -1 if no filters */
p_input->input_alloc.i_bytes_per_sec = __MAX(
p_input->input_alloc.i_bytes_per_sec,
- p_input->input.i_bytes_per_frame
+ (int)(p_input->input.i_bytes_per_frame
* p_input->input.i_rate
- / p_input->input.i_frame_length );
+ / p_input->input.i_frame_length) );
/* Allocate in the heap, it is more convenient for the decoder. */
p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
aout_buffer_t * p_buffer )
{
- mtime_t start_date, duration;
+ mtime_t start_date;
/* We don't care if someone changes the start date behind our back after
* this. We'll deal with that when pushing the buffer, and compensate
/* The decoder is _very_ late. This can only happen if the user
* pauses the stream (or if the decoder is buggy, which cannot
* happen :). */
- msg_Warn( p_aout, "computed PTS is out of range (%lld), clearing out",
- start_date );
- vlc_mutex_lock( &p_aout->mixer_lock );
+ msg_Warn( p_aout, "computed PTS is out of range ("I64Fd"), "
+ "clearing out", mdate() - start_date );
+ vlc_mutex_lock( &p_aout->input_fifos_lock );
aout_FifoSet( p_aout, &p_input->fifo, 0 );
- vlc_mutex_unlock( &p_aout->mixer_lock );
+ vlc_mutex_unlock( &p_aout->input_fifos_lock );
+ if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
+ msg_Warn( p_aout, "timing screwed, stopping resampling" );
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+ if ( p_input->i_nb_resamplers != 0 )
+ {
+ p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ p_input->pp_resamplers[0]->b_reinit = VLC_TRUE;
+ }
start_date = 0;
- }
+ }
if ( p_buffer->start_date < 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 (%lld), dropping buffer",
+ msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer",
mdate() - p_buffer->start_date );
aout_BufferFree( p_buffer );
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
- if ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
- || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE )
+ /* 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 ) &&
+ p_input->i_nb_resamplers > 0 )
{
/* Can happen in several circumstances :
* 1. A problem at the input (clock drift)
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
- int i_ratio;
- mtime_t old_duration;
mtime_t drift = p_buffer->start_date - start_date;
- msg_Warn( p_aout, "buffer is %lld %s, resampling",
- drift > 0 ? drift : -drift,
- drift > 0 ? "in advance" : "late" );
- old_duration = p_buffer->end_date - p_buffer->start_date;
- duration = p_buffer->end_date - start_date;
- i_ratio = (duration * 100) / old_duration;
- /* If the ratio is too != 100, the sound quality will be awful. */
- if ( i_ratio < 100 - AOUT_MAX_RESAMPLING /* % */ )
+ p_input->i_resamp_start_date = mdate();
+ p_input->i_resamp_start_drift = (int)drift;
+
+ if ( drift > 0 )
+ p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
+ else
+ p_input->i_resampling_type = AOUT_RESAMPLING_UP;
+
+ msg_Warn( p_aout, "buffer is "I64Fd" %s, triggering %ssampling",
+ drift > 0 ? drift : -drift,
+ drift > 0 ? "in advance" : "late",
+ drift > 0 ? "down" : "up");
+ }
+
+ if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
+ {
+ /* Resampling has been triggered previously (because of dates
+ * mismatch). We want the resampling to happen progressively so
+ * it isn't too audible to the listener. */
+
+ if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
{
- duration = (old_duration * (100 - AOUT_MAX_RESAMPLING)) / 100;
+ p_input->pp_resamplers[0]->input.i_rate += 10; /* Hz */
}
- if ( i_ratio > 100 + AOUT_MAX_RESAMPLING /* % */ )
+ else
{
- duration = (old_duration * (100 + AOUT_MAX_RESAMPLING)) / 100;
+ p_input->pp_resamplers[0]->input.i_rate -= 10; /* Hz */
}
- p_input->pp_resamplers[0]->input.i_rate
- = (p_input->input.i_rate * old_duration) / duration;
- aout_FiltersPlay( p_aout, p_input->pp_resamplers,
- p_input->i_nb_resamplers,
- &p_buffer );
- }
- else
- {
- duration = p_buffer->end_date - p_buffer->start_date;
-
- if ( p_input->input.i_rate != p_aout->mixer.mixer.i_rate )
+ /* Check if everything is back to normal, in which case we can stop the
+ * resampling */
+ if( p_input->pp_resamplers[0]->input.i_rate ==
+ p_input->input.i_rate )
+ {
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
+ msg_Warn( p_aout, "resampling stopped after "I64Fi" usec",
+ mdate() - p_input->i_resamp_start_date );
+ }
+ else if( abs( (int)(p_buffer->start_date - start_date) ) <
+ abs( p_input->i_resamp_start_drift ) / 2 )
+ {
+ /* if we reduced the drift from half, then it is time to switch
+ * back the resampling direction. */
+ if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
+ p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
+ else
+ p_input->i_resampling_type = AOUT_RESAMPLING_UP;
+ p_input->i_resamp_start_drift = 0;
+ }
+ else if( p_input->i_resamp_start_drift &&
+ ( abs( (int)(p_buffer->start_date - start_date) ) >
+ abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
{
- /* Standard resampling is needed ! */
+ /* If the drift is increasing and not decreasing, than something
+ * is bad. We'd better stop the resampling right now. */
+ msg_Warn( p_aout, "timing screwed, stopping resampling" );
+ p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
-
- aout_FiltersPlay( p_aout, p_input->pp_resamplers,
- p_input->i_nb_resamplers,
- &p_buffer );
}
}
/* 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 = start_date + duration;
+
+ /* Actually run the resampler now. */
+ if ( p_input->i_nb_resamplers > 0 && p_aout->mixer.mixer.i_rate !=
+ p_input->pp_resamplers[0]->input.i_rate )
+ {
+ aout_FiltersPlay( p_aout, p_input->pp_resamplers,
+ p_input->i_nb_resamplers,
+ &p_buffer );
+ }
vlc_mutex_lock( &p_aout->input_fifos_lock );
aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
projects / vlc / blobdiff