int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,                    aout_buffer_t * p_buffer ){    mtime_t start_date;    if( p_input->b_restart )    {        aout_fifo_t fifo, dummy_fifo;        byte_t      *p_first_byte_to_mix;        vlc_mutex_lock( &p_aout->mixer_lock );        /* A little trick to avoid loosing our input fifo */        aout_FifoInit( p_aout, &dummy_fifo, p_aout->mixer.mixer.i_rate );        p_first_byte_to_mix = p_input->p_first_byte_to_mix;        fifo = p_input->fifo;        p_input->fifo = dummy_fifo;        aout_InputDelete( p_aout, p_input );        aout_InputNew( p_aout, p_input );        p_input->p_first_byte_to_mix = p_first_byte_to_mix;        p_input->fifo = fifo;        vlc_mutex_unlock( &p_aout->mixer_lock );    }    /* 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. */    vlc_mutex_lock( &p_aout->input_fifos_lock );    start_date = aout_FifoNextStart( p_aout, &p_input->fifo );    vlc_mutex_unlock( &p_aout->input_fifos_lock );    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 ("I64Fd"), "                  "clearing out", mdate() - start_date );        vlc_mutex_lock( &p_aout->input_fifos_lock );        aout_FifoSet( p_aout, &p_input->fifo, 0 );        p_input->p_first_byte_to_mix = NULL;        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_continuity = VLC_FALSE;        }        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 ("I64Fd"), dropping buffer",                  mdate() - p_buffer->start_date );        aout_BufferFree( p_buffer );        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_continuity = VLC_FALSE;        }        return 0;    }    /* If the audio drift is too big then it's not worth trying to resample     * the audio. */    if ( start_date != 0 &&         ( start_date < p_buffer->start_date - 3 * AOUT_PTS_TOLERANCE ) )    {        msg_Warn( p_aout, "audio drift is too big ("I64Fd"), clearing out",                  start_date - p_buffer->start_date );        vlc_mutex_lock( &p_aout->input_fifos_lock );        aout_FifoSet( p_aout, &p_input->fifo, 0 );        p_input->p_first_byte_to_mix = NULL;        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_continuity = VLC_FALSE;        }        start_date = 0;    }    else if ( start_date != 0 &&              ( start_date > p_buffer->start_date + 3 * AOUT_PTS_TOLERANCE ) )    {        msg_Warn( p_aout, "audio drift is too big ("I64Fd"), dropping buffer",                  start_date - p_buffer->start_date );        aout_BufferFree( p_buffer );        return 0;    }    if ( start_date == 0 ) start_date = p_buffer->start_date;    /* Run pre-filters. */    aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,                      &p_buffer );    /* 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)         * 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->start_date - 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 "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 )        {            p_input->pp_resamplers[0]->input.i_rate += 2; /* Hz */        }        else        {            p_input->pp_resamplers[0]->input.i_rate -= 2; /* Hz */        }        /* 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 "                      "(drift: "I64Fi")",                      mdate() - p_input->i_resamp_start_date,                      p_buffer->start_date - 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 ) )        {            /* 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;        }    }    /* 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;    /* Actually run the resampler now. */    if ( p_input->i_nb_resamplers > 0 )    {        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 );    vlc_mutex_unlock( &p_aout->input_fifos_lock );    return 0;}static int ChangeFiltersString( aout_instance_t * p_aout,                                 char *psz_name, vlc_bool_t b_add ){    vlc_value_t val;    char *psz_parser;    var_Get( p_aout, "audio-filter", &val );    if( !val.psz_string ) val.psz_string = strdup("");    psz_parser = strstr( val.psz_string, psz_name );    if( b_add )    {        if( !psz_parser )        {            psz_parser = val.psz_string;            asprintf( &val.psz_string, (*val.psz_string) ? "%s:%s" : "%s%s",                      val.psz_string, psz_name );            free( psz_parser );        }        else        {            return 0;        }    }    else    {        if( psz_parser )        {            memmove( psz_parser, psz_parser + strlen(psz_name) +                     (*(psz_parser + strlen(psz_name)) == ':' ? 1 : 0 ),                     strlen(psz_parser + strlen(psz_name)) + 1 );        }        else        {            free( val.psz_string );            return 0;        }    }    var_Set( p_aout, "audio-filter", val );    free( val.psz_string );    return 1;}static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd,                       vlc_value_t oldval, vlc_value_t newval, void *p_data ){    aout_instance_t *p_aout = (aout_instance_t *)p_this;    char *psz_mode = newval.psz_string;    vlc_value_t val;    int i;    if( !psz_mode || !*psz_mode )    {        ChangeFiltersString( p_aout, "goom", VLC_FALSE );        ChangeFiltersString( p_aout, "visual", VLC_FALSE );        ChangeFiltersString( p_aout, "galaktos", VLC_FALSE );    }    else    {        if( !strcmp( "goom", psz_mode ) )        {            ChangeFiltersString( p_aout, "visual", VLC_FALSE );            ChangeFiltersString( p_aout, "goom", VLC_TRUE );            ChangeFiltersString( p_aout, "galaktos", VLC_FALSE );        }        else if( !strcmp( "galaktos", psz_mode ) )        {            ChangeFiltersString( p_aout, "visual", VLC_FALSE );            ChangeFiltersString( p_aout, "goom", VLC_FALSE );            ChangeFiltersString( p_aout, "galaktos", VLC_TRUE );        }        else        {            val.psz_string = psz_mode;            var_Create( p_aout, "effect-list", VLC_VAR_STRING );            var_Set( p_aout, "effect-list", val );            ChangeFiltersString( p_aout, "goom", VLC_FALSE );            ChangeFiltersString( p_aout, "visual", VLC_TRUE );            ChangeFiltersString( p_aout, "galaktos", VLC_FALSE );        }    }    /* That sucks */    for( i = 0; i < p_aout->i_nb_inputs; i++ )    {        p_aout->pp_inputs[i]->b_restart = VLC_TRUE;    }    return VLC_SUCCESS;}static int EqualizerCallback( vlc_object_t *p_this, char const *psz_cmd,                       vlc_value_t oldval, vlc_value_t newval, void *p_data ){    aout_instance_t *p_aout = (aout_instance_t *)p_this;    char *psz_mode = newval.psz_string;    vlc_value_t val;    int i;    int i_ret;    if( !psz_mode || !*psz_mode )    {        i_ret = ChangeFiltersString( p_aout, "equalizer", VLC_FALSE );    }    else    {        val.psz_string = psz_mode;        var_Create( p_aout, "equalizer-preset", VLC_VAR_STRING );        var_Set( p_aout, "equalizer-preset", val );        i_ret = ChangeFiltersString( p_aout, "equalizer", VLC_TRUE );    }    /* That sucks */    if( i_ret == 1 )    {        for( i = 0; i < p_aout->i_nb_inputs; i++ )        {            p_aout->pp_inputs[i]->b_restart = VLC_TRUE;        }    }    return VLC_SUCCESS;}static aout_filter_t * allocateUserChannelMixer( aout_instance_t * p_aout,                                     audio_sample_format_t * p_input_format,                                     audio_sample_format_t * p_output_format ){    aout_filter_t * p_channel_mixer;    /* Retreive user preferred channel mixer */    char * psz_name = config_GetPsz( p_aout, "audio-channel-mixer" );    /* Not specified => let the main pipeline do the mixing */    if ( ! psz_name ) return NULL;    /* Debug information */    aout_FormatsPrint( p_aout, "channel mixer", p_input_format,                       p_output_format );    /* Create a VLC object */    p_channel_mixer = vlc_object_create( p_aout, sizeof(aout_filter_t) );    if( p_channel_mixer == NULL )    {        msg_Err( p_aout, "cannot add user channel mixer %s", psz_name );        return NULL;    }    vlc_object_attach( p_channel_mixer , p_aout );    /* Attach the suitable module */    memcpy( &p_channel_mixer->input, p_input_format,                    sizeof(audio_sample_format_t) );    memcpy( &p_channel_mixer->output, p_output_format,                    sizeof(audio_sample_format_t) );    p_channel_mixer->p_module =        module_Need( p_channel_mixer,"audio filter", psz_name, VLC_TRUE );    if( p_channel_mixer->p_module== NULL )    {        msg_Err( p_aout, "cannot add user channel mixer %s", psz_name );        vlc_object_detach( p_channel_mixer );        vlc_object_destroy( p_channel_mixer );        return NULL;    }    p_channel_mixer->b_continuity = VLC_FALSE;    /* Ok */    return p_channel_mixer;}