/*****************************************************************************
* aout_InputCheckAndRestart : restart an input
*****************************************************************************
* This function must be entered with the input and mixer lock.
*****************************************************************************/
void aout_InputCheckAndRestart( aout_instance_t * p_aout, aout_input_t * p_input )
{
AOUT_ASSERT_MIXER_LOCKED;
AOUT_ASSERT_INPUT_LOCKED;
if( !p_input->b_restart )
return;
aout_lock_input_fifos( p_aout );
/* A little trick to avoid loosing our input fifo and properties */
uint8_t *p_first_byte_to_mix = p_input->mixer.begin;
aout_fifo_t fifo = p_input->mixer.fifo;
bool b_paused = p_input->b_paused;
mtime_t i_pause_date = p_input->i_pause_date;
aout_FifoInit( p_aout, &p_input->mixer.fifo, p_aout->mixer_format.i_rate );
aout_InputDelete( p_aout, p_input );
aout_InputNew( p_aout, p_input, &p_input->request_vout );
p_input->mixer.begin = p_first_byte_to_mix;
p_input->mixer.fifo = fifo;
p_input->b_paused = b_paused;
p_input->i_pause_date = i_pause_date;
p_input->b_restart = false;
aout_unlock_input_fifos( p_aout );
}
//#define AOUT_PROCESS_BEFORE_CHEKS
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
aout_buffer_t * p_buffer, int i_input_rate )
{
mtime_t start_date;
AOUT_ASSERT_INPUT_LOCKED;
if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL )
{
inputDrop( p_input, p_buffer );
return 0;
}
#ifdef AOUT_PROCESS_BEFORE_CHEKS
/* Run pre-filters. */
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
if( !p_buffer )
return 0;
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
{
const mtime_t i_date = p_buffer->i_pts;
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
}
if( !p_buffer )
return 0;
if( p_buffer->i_nb_samples <= 0 )
{
block_Release( p_buffer );
return 0;
}
#endif
/* Handle input rate change, but keep drift correction */
if( i_input_rate != p_input->i_last_input_rate )
{
unsigned int * const pi_rate = &p_input->p_playback_rate_filter->fmt_in.audio.i_rate;
#define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) )
const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate);
*pi_rate = F(p_input->input.i_rate + i_delta, i_input_rate);
#undef F
p_input->i_last_input_rate = i_input_rate;
}
/* 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
* with the next incoming buffer. */
aout_lock_input_fifos( p_aout );
start_date = aout_FifoNextStart( p_aout, &p_input->mixer.fifo );
aout_unlock_input_fifos( p_aout );
if ( start_date != 0 && start_date < mdate() )
{
/* 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 (%"PRId64"), "
"clearing out", mdate() - start_date );
aout_lock_input_fifos( p_aout );
aout_FifoSet( p_aout, &p_input->mixer.fifo, 0 );
p_input->mixer.begin = NULL;
aout_unlock_input_fifos( p_aout );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
start_date = 0;
}
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->i_pts );
inputDrop( p_input, p_buffer );
inputResamplingStop( p_input );
return 0;
}
/* If the audio drift is too big then it's not worth trying to resample
* 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->i_pts - i_pts_tolerance ) )
{
msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out",
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;
aout_unlock_input_fifos( p_aout );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
start_date = 0;
}
else if ( start_date != 0 &&
( 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->i_pts );
inputDrop( p_input, p_buffer );
return 0;
}
if ( start_date == 0 ) start_date = p_buffer->i_pts;
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* Run pre-filters. */
aout_FiltersPlay( p_input->pp_filters, p_input->i_nb_filters, &p_buffer );
if( !p_buffer )
return 0;
#endif
/* 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->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 :
* 1. A problem at the input (clock drift)
* 2. A small pause triggered by the user
* 3. Some delay in the output stage, causing a loss of lip
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
mtime_t drift = p_buffer->i_pts - start_date;
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 %"PRId64" %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 )
{
p_input->pp_resamplers[0]->fmt_in.audio.i_rate += 2; /* Hz */
}
else
{
p_input->pp_resamplers[0]->fmt_in.audio.i_rate -= 2; /* Hz */
}
/* Check if everything is back to normal, in which case we can stop the
* resampling */
unsigned int i_nominal_rate =
(p_input->pp_resamplers[0] == p_input->p_playback_rate_filter)
? INPUT_RATE_DEFAULT * p_input->input.i_rate / i_input_rate
: p_input->input.i_rate;
if( p_input->pp_resamplers[0]->fmt_in.audio.i_rate == i_nominal_rate )
{
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec "
"(drift: %"PRIi64")",
mdate() - p_input->i_resamp_start_date,
p_buffer->i_pts - 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
* 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->i_pts - start_date) ) >
abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
{
/* 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" );
inputResamplingStop( p_input );
p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
}
}
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
{
aout_FiltersPlay( p_input->pp_resamplers, p_input->i_nb_resamplers,
&p_buffer );
}
if( !p_buffer )
return 0;
if( p_buffer->i_nb_samples <= 0 )
{
block_Release( p_buffer );
return 0;
}
#endif
/* Adding the start date will be managed by aout_FifoPush(). */
p_buffer->i_pts = start_date;
aout_lock_input_fifos( p_aout );
aout_FifoPush( p_aout, &p_input->mixer.fifo, p_buffer );
aout_unlock_input_fifos( p_aout );
return 0;
}
/*****************************************************************************
* aout_Restart : re-open the output device and rebuild the input and output
* pipelines
*****************************************************************************
* This function is used whenever the parameters of the output plug-in are
* changed (eg. selecting S/PDIF or PCM).
*****************************************************************************/
static int aout_Restart( aout_instance_t * p_aout )
{
int i;
bool b_error = 0;
aout_lock_mixer( p_aout );
if ( p_aout->i_nb_inputs == 0 )
{
aout_unlock_mixer( p_aout );
msg_Err( p_aout, "no decoder thread" );
return -1;
}
/* Lock all inputs. */
aout_lock_input_fifos( p_aout );
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
aout_lock_input( p_aout, p_aout->pp_inputs[i] );
aout_InputDelete( p_aout, p_aout->pp_inputs[i] );
}
aout_MixerDelete( p_aout );
/* Re-open the output plug-in. */
aout_OutputDelete( p_aout );
if ( aout_OutputNew( p_aout, &p_aout->pp_inputs[0]->input ) == -1 )
{
/* Release all locks and report the error. */
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
vlc_mutex_unlock( &p_aout->pp_inputs[i]->lock );
}
aout_unlock_input_fifos( p_aout );
aout_unlock_mixer( p_aout );
return -1;
}
if ( aout_MixerNew( p_aout ) == -1 )
{
aout_OutputDelete( p_aout );
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
vlc_mutex_unlock( &p_aout->pp_inputs[i]->lock );
}
aout_unlock_input_fifos( p_aout );
aout_unlock_mixer( p_aout );
return -1;
}
/* Re-open all inputs. */
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
aout_input_t * p_input = p_aout->pp_inputs[i];
b_error |= aout_InputNew( p_aout, p_input );
p_input->b_changed = 1;
aout_unlock_input( p_aout, p_input );
}
aout_unlock_input_fifos( p_aout );
aout_unlock_mixer( p_aout );
return b_error;
}
/*****************************************************************************
* doVolumeChanges : handle all volume changes. Internal use only to ease
* variables locking.
*****************************************************************************/
static
int doVolumeChanges( unsigned action, vlc_object_t * p_object, int i_nb_steps,
audio_volume_t i_volume, audio_volume_t * i_return_volume,
bool b_mute )
{
int i_result = VLC_SUCCESS;
int i_volume_step = 1, i_new_volume = 0;
bool b_var_mute = false;
aout_instance_t *p_aout = findAout( p_object );
if ( p_aout ) aout_lock_volume( p_aout );
b_var_mute = var_GetBool( p_object, "volume-muted");
const bool b_unmute_condition = ( b_var_mute
&& ( /* Unmute: on increments */
( action == INCREMENT_VOLUME )
|| /* On explicit unmute */
( ( action == SET_MUTE ) && !b_mute )
|| /* On toggle from muted */
( action == TOGGLE_MUTE )
));
const bool b_mute_condition = ( !b_var_mute
&& ( /* explicit */
( ( action == SET_MUTE ) && b_mute )
|| /* or toggle */
( action == TOGGLE_MUTE )
));
/* If muting or unmuting when play hasn't started */
if ( action == SET_MUTE && !b_unmute_condition && !b_mute_condition )
{
if ( p_aout )
{
aout_unlock_volume( p_aout );
vlc_object_release( p_aout );
}
return i_result;
}
/* On UnMute */
if ( b_unmute_condition )
{
/* Restore saved volume */
i_volume = var_GetInteger( p_object, "saved-volume" );
var_SetBool( p_object, "volume-muted", false );
}
else if ( b_mute_condition )
{
/* We need an initial value to backup later */
i_volume = config_GetInt( p_object, "volume" );
}
if ( action == INCREMENT_VOLUME )
{
i_volume_step = var_InheritInteger( p_object, "volume-step" );
if ( !b_unmute_condition )
i_volume = config_GetInt( p_object, "volume" );
i_new_volume = (int) i_volume + i_volume_step * i_nb_steps;
if ( i_new_volume > AOUT_VOLUME_MAX )
i_volume = AOUT_VOLUME_MAX;
else if ( i_new_volume < AOUT_VOLUME_MIN )
i_volume = AOUT_VOLUME_MIN;
else
i_volume = i_new_volume;
}
var_SetInteger( p_object, "saved-volume" , i_volume );
/* On Mute */
if ( b_mute_condition )
{
i_volume = AOUT_VOLUME_MIN;
var_SetBool( p_object, "volume-muted", true );
}
/* Commit volume changes */
config_PutInt( p_object, "volume", i_volume );
if ( p_aout )
{
aout_lock_mixer( p_aout );
aout_lock_input_fifos( p_aout );
if ( p_aout->p_mixer )
i_result = p_aout->output.pf_volume_set( p_aout, i_volume );
aout_unlock_input_fifos( p_aout );
aout_unlock_mixer( p_aout );
}
/* trigger callbacks */
var_TriggerCallback( p_object, "volume-change" );
if ( p_aout )
{
var_SetBool( p_aout, "intf-change", true );
aout_unlock_volume( p_aout );
vlc_object_release( p_aout );
}
if ( i_return_volume != NULL )
*i_return_volume = i_volume;
return i_result;
}
